[Asterisk-Users] voipcheap.com - miracle free land line calls
Hi All,
Someone can explain how that miracle free landline calls is made?
I´ve tried this with my server and it works, but...how they do it?
Miklos
IPFONE TELEFONIA IP
Rua Caio Graco 735 São Paulo SP
IPBX - +55 11 3488-3800
http://www.ipfone.com.br
[EMAIL PROTECTED]
Balbus balbum intellegit
- Original Message -
From: Marco Mouta [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 07, 2006 8:03 PM
Subject: Re: [Asterisk-Users] meetme public
Hi,
Please check you [general] section in sip.conf
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying The number you have dialed is not in service. Please check the
; number and try again.
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
It could be happening that your public sip call is arriving @ asterisk,
and
seems unknow, so it is sent to from-sip-external context.
In your extensions.conf look for section called [from-sip-external], there
you need to paste your code to route the call to your meetme room.
Hope it helps,
Best regards,
Marco Mouta
Ps. Please give me some feeback if it solved.
On 6/7/06, Pablo Allietti [EMAIL PROTECTED] wrote:
hi all i have an asterisk working and i need to add a mettme public
service.
for example i need to download a soft (sjphone) and without any
configuration call to [EMAIL PROTECTED] (meetme) and join a
conference but when i do that i
received an error saying nomber do not exist. but if i call a extension
is work propperly.
in the extensions.conf have
exten = 411,1,Answer
exten = 411,2,Wait(1)
exten =
411,3,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP})
exten = 411,4,Monitor(wav,${TIMESTAMP},m)
exten = 411,5,Meetme(4001,qM)
exten = 411,6,Hangup
4001 is the room number
in the mmetme conf have
conf = 4001
any comments?
--
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Com os melhores cumprimentos,
Marco Mouta
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[Asterisk-Users] USING MMS STREAM FOR MOH
Hi All, I need to use - mms://61.112.173.60:81/ as souce for MOH, i cant find anything about using that souce format in wiki. If you have some info please advice. Miklos IPFONE TELEFONIA IP Rua Caio Graco 735 São Paulo SP IPBX - +55 11 3488-3800 http://www.ipfone.com.br [EMAIL PROTECTED] Balbus balbum intellegit - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 4:06 PM Subject: RE: [Asterisk-Users] how to show called name on calling polycomdisplay Why? If you flip the callerid and dnis variables, it should work with any phone. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 15, 2006 11:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] how to show called name on calling polycomdisplay I will test it, However it is still PolyCom Specific. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bowyer Sent: Wednesday, March 15, 2006 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how to show called name on calling polycomdisplay On 3/15/06, Alexander Lopez [EMAIL PROTECTED] wrote: This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. I guess the developers that have worked on implementing this should be told that, then. http://bugs.digium.com/view.php?id=6643 This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Afana Sent: Wednesday, March 15, 2006 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how to show called name on calling polycomdisplay I was looking for this exactly as well Any ideas? - Gabe - Original Message - From: Giorgio Incantalupo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 15, 2006 12:52 AM Subject: [Asterisk-Users] how to show called name on calling polycom display Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EDGE-CORE SIP PHONE
Hi All! I need some feedback about the edge-core sip phones, somebody uses it? They are reliable? What the community say about them? Miklos ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wav to g729
Try the new conversion module from redice li ..it is greate! Miklos IPFONE TELEFONIA IP Rua Caio Graco 735 São Paulo SP IPBX - +55 11 3488-3800 http://www.ipfone.com.br [EMAIL PROTECTED] Balbus balbum intellegit - Original Message - From: Innocent Evil [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 22, 2005 5:00 PM Subject: Re: [Asterisk-Users] wav to g729 I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 19:44:36 +0100 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http://www.asteriskguru.com/tools/audio_conversion.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstLinux 0.2.9 Released
Hi Kristian, I installed 0.2.9 today ..it is grate...the zaptel / ztdummy issues are gone an the systems are going very well. Thanks and congratulations for the always good work. Have you seem that new grafical interface using ruby? maybe it can be integrated in astlinux...what you think about? Sorry if i´m saying something stupid... Regards Miklos IPFONE TELEFONIA IP Rua Caio Graco 735 São Paulo SP IPBX - +55 11 3488-3800 http://www.ipfone.com.br [EMAIL PROTECTED] Balbus balbum intellegit - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 21, 2005 5:09 PM Subject: Re: [Asterisk-Users] AstLinux 0.2.9 Released Ben Higley wrote: this would be very beneficial to me as well.. I have the S518 ADSL card in my Linux system as well.. I was looking at going to an ASTLINUX solution. Hi Kristian, Excellent thanks.. On 11/21/05, Kristian Kielhofner [EMAIL PROTECTED] wrote: Hello Everyone, I have finished up work on what will (hopefully) become AstLinux 0.3.0. AstLinux 0.2.9 has been released as a test release, and includes the following changes: - Asterisk 1.2.0 - Zaptel 1.2.0 - libpri 1.2.0 - Sangoma wanrouter beta1-2.3.4 Does this mean the Sangoma S518 ADSL Card may work on Astlinux on a soekris 4810 board do you know? thanks Mike Support for the Sangoma S518 ADSL card has been in AstLinux for a long time. Wanpipe/wanrouter has just been upgraded to a newer version. And yes, the S518 does work on the Soekris net4801. That's what I have at home. -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.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] VIDEO ON 1.0.7 stable
Thanks! it works very well, only to inform other brazilians: No problems to send video in an embratel dedicated 1mb link from a tva 512kbs. miklos - Original Message - From: Erdem HAKI [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 26, 2005 9:31 AM Subject: Re: [Asterisk-Users] VIDEO ON 1.0.7 stable - Original Message - From: Nardis Dome [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 26, 2005 1:59 PM Subject: Re: [Asterisk-Users] VIDEO ON 1.0.7 stable --- listas iPfone [EMAIL PROTECTED] wrote: Hi all I need to know if the video support for h.263 is active in version stable 1.0.7 to use with eyeBeam in asterisk it works for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks Nardis Dome, it shows the way. I use [EMAIL PROTECTED] ,eyeBeam video feature on asterisk didn't work first, but after adding allow=h263p , it has worked properly. [2001] username=2001 type=friend secret=** qualify=no port=5060 pickupgroup= nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all context=from-internal canreinvite=no callgroup= callerid=Erdem HAKI 2001 allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 allow=h263p ___ 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 x PROLIANT ML 150 G2 SATA
Hi All, I´m tryingo to install asterisk in an PROLIANT ML 150 G2 SATA and can´t make it work because linux cant recognize the Hd (HP 160 mb). No drivers for Centos ...Red Hat... i´t´s drivig me crazy.. Someone have a tip? if i make change it to SCSI i think it will work but not sure about. Thanks Miklos ___ 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] VIDEO ON 1.0.7 stable
Hi all I need to know if the video support for h.263 is active in version stable 1.0.7 to use with eyeBeam in asterisk In the wiki the info is that this support is from CVS HEAD 02/25/2005 Thanks Miklos ___ 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] IP500 Registration
Did you checked the outbound proxy parameter? - Original Message - From: David Sampson [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Wednesday, May 04, 2005 4:05 PM Subject: [Asterisk-Users] IP500 Registration Hello - I have an IP500 (my first). The phone is up and running and I am able to make outgoing calls but I can't get the phone to register and take incoming calls. This is what my sip.conf looks like: [8503] type=user username=dave callerid=Dave Sampson 8503 secret=default host=dynamic dtmfmode=inband context=millenium mailbox=8503 defaultip=10.10.5.53 progressinband=no SIP debug shows: May 4 14:57:51 NOTICE[10797]: chan_sip.c:7691 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '10.10.5.53' Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[EMAIL PROTECTED]' Any help is greatly appreciated. No NAT here - just on the private net. Thanks, Dave ___ 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] *ASTERISK* Install and configure Step by Step.
Hi Max! I am Begining in the ASTERISK IP-PABX world, and here in Brazil, have not any Help to install and configure, Sure you have!: http://www.ipfone.com.br/curso.asp Miklos - Original Message - From: Max [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 01, 2005 8:36 PM Subject: [Asterisk-Users] *ASTERISK* Install and configure Step by Step. Hello! I am Begining in the ASTERISK IP-PABX world, and here in Brazil, have not any Help to install and configure, If you know about any Good LINK contend HOW TO install and configure Asterisk to this hardware(minimal) OR if exist mini linux distro run asterisk in RAM, (similar at coyotelinux.com) bienvenidas todas las ideas! INTEL MMX CPU 166Mhz 32MB Ram HD 20GB Lan cart 10/100Mb Fax modem genius (Lucent chipset) Fax Modem USR 33.66 Sound OnBoard Disk Driver 1.44 CD 52X I need Send to my PABX, using only 1 FXS port all incoming Calls from Internet I have multiple SIP servers and providers(6 ip lines, vitual numbers) this is Posible using asterisk? Thanks in advace, Max Rivera ___ 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] Linksys PAP2-NA Config
Hi! Use the spa2000 configuration info, the software is the same. Miklos - Original Message - From: Listas [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 23, 2004 4:46 PM Subject: Re: [Asterisk-Users] Linksys PAP2-NA Config Ok I forgot to ask if any of you out there have fought against any of this issues and have any information that can (and has the will) to share... or if any of you has any kind of documentation about this. thanks again, Matias - Original Message - From: Listas To: asterisk-users@lists.digium.com Sent: Thursday, December 23, 2004 12:13 PM Subject: [Asterisk-Users] Linksys PAP2-NA Config Hi, I have 3 PAP2 connected to *, they work fine but there are some things which I would like to improve, some of them are: - double ring tone when placing a call (I hear two tones it seems like the PAP2 is generating it's own tone) - some kind of noise (like glitches or something) when I pick up the phone (seems like some polarity thing) - I'd like to keep the tone after pressing 9 (just like ignorepat in *, but in the PAP2 dialplan) there are so many options in the PAP2 that I haven't been able to achieve this things, I'm aware that maybe some of them are not possible, but I couldn't find any documentation on configuring the PAP2-NA... thanks in advance. Matias -- ___ 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] snom 200 and * question
Hi all I have question regarding to my nom 200 and asterisk. I have an * server with two x101p and two lines conected. When i am in a call in line 1 and a call in line two is received the first call goes imediatly to hold and the line button blinks indicating that another call arrived. It is very bad because i can,t inform the first caller that the another call is waiting... i have to take the second call tell the second caller to wait and return to the first call. What can i do to make the first call remain until i attend the second? Snom 200 runing software 2.03o Asterisk from CVS 01/21/04 dialplan: When the caller press 2 or 3 [sales] exten = s,1,SetCallerID(2) exten = s,2,SetCIDName(sales) exten = s,3,DIAL(SIP/snom200,20,tr) exten = s,4,Wait(1) exten = s,5,Goto(recepcao,s,1) [support] exten = s,1,SetCallerID(3) exten = s,2,SetCIDName(suporte) exten = s,3,DIAL(SIP/snom200,20,tr) exten = s,4,Wait(1) exten = s,5,Goto(recepcao,s,1) Thanks for any help. Atenciosamente Cláudio Miklos iPFONE Telefonia IP Rua Caio Graco 735 São Paulo SP BR - 55 11 3801-3702 USA - 1 360-968-1591 FWD - 64662 sip:[EMAIL PROTECTED] www.ipfone.com.br [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FIREFLY repeat calls
Hi! I´m trying to use firefly 3 party with * and iax2. I cant figure out why it reapeats every call many times until it is closed. It is a bug ? I want it because of the skin changing thing.. Someone have a clue on how to use it with * Thanks Miklos - Original Message - From: Jozeph Brasil [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 11:17 AM Subject: RES: [Asterisk-Users] Softphone - Freeware?! I have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a Flash+Extension+Hangup CALL... Thanks for all! -Mensagem original- De: Eric Bart [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 10:51 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! axra will do. it's an add-on that will give consultative transfer to X-Lite (and others). see below : --- New application for asterisk : axra axra runs separately. developped in C++. it dialogs with asterisk through agi calls and through the manager api. it proccesses phone calls through the dial plan (agi) and concurently through the manager api. axra currently provides consultative transfer for SIP and IAX2 phones. this should easily be extended to any phone technology. hopefully, axra will soon provide 3 way calling. there are two tranfer functions : PreTransfer and CTransfer each should be implemented in the dial plan like : exten = 76,1,AGI(axraagi|PreTransfer) exten = 76,2,Hangup exten = 77,1,AGI(axraagi|CTransfer|auto) exten = 77,2,Hangup you may choose other extensions than 76 77. you may omit 'auto' when a call is transfered to PreTransfer (76), the call is parked and waits for a transfer. if the timeout occurs, the call is ringed back. if you call PreTransfer (76) directly, the parked call (if any) is immediatly ringed back. when a call is transfered to CTransfer (77), the call is linked to the pretransfered (parked) call. if no pretransfer exists the call is pretransfered just like 76 was dialed. however, if 'auto' was specified, axra will try to link the call to the oldest live channel attached to transferer's phone. http://www.byortek.com/asterisk/axra-2004-07-29.tgz Please download, read REAME and INSTALL. Any feedback greatly appreciated. - Original Message - From: Jozeph Brasil [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 3:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! Hi everybody, What is the most complete Softphone application freeware? X-Lite is very CooL, but the free version don´t support transfers... :( Anyone know, a windows softphone free application that I can use all Askterisk Resources? Congratulations, Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP Soundpoint 600 early dial
Hi! You can do this in the web interface sip conf local settings Digitmap You can map the number of digits to be dialed before sending..etc... miklos - Original Message - From: Tor Setane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 9:26 AM Subject: RE: [Asterisk-Users] Polycom IP Soundpoint 600 early dial Mike Roberts wrote: Is anyone successfully using this phone with *? I have one, and it is an excellent phone. However, I cannot figure out how to make the phone early dial -- that is, automatically dial the number without the user having to press the send button. Any ideas? Thanks, Mike Roberts If you access the phone with a web browser, you can add a digitmap in Sip Conf - Local Settings If you have four digit internal numbers, 0 for operator, 9 for outside line: 0[1-8]xxx|9,T etc. The comma just gives you a new dialtone, and the T waits for the timeout you choose as Digitmap Timeout on the same page. This is just an example, you would probably be better off building a more complete digitmap. Regards, Tor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] crazy optipoint 400sip and asterisk
Hi all, after a good time trying i made the optipoint work with asterisk... this is very strange but.. maybe someone can do it and tell me what happens: I have two peers in sip.conf : [19] accountcode=19 amaflags=billing type=friend username=19 secret= host=dynamic nat=yes qualify=1000 context=sip [optipoint] type=friend secret= username=optipoint host=dynamic defaultip=192.168.0.36 dtmfmode=inband canreinvite=yes nat=yes qualify=1000 mailbox=331 context=internal callerid=optipoint optipoint In extensions.conf: exten = 37,1,DIAL(SIP/optipoint,20,tr) The optipoint is configured in that way: Terminal number: 19 Terminal name: 19 Register by terminal name: x SIP details: SIP routing: Gateway Registrar IP address or DNS name: 192.168.0.34 (my *) Server IP address or DNS name: 192.168.0.34 Gateway IP address or DNS name: 192.168.0.34 Outbound proxy: Default OBP domain: SIP transport: UDP SIP session timer enabled: x SIP session timer value (1800-3600): 3600seconds Registration timer value: 3600seconds SIP realm: asterisk SIP user ID: 19 New SIP password: - Confirm SIP password: - Beep on SIP server error: x cli shows: asterisk-eth0*CLI sip show peers Name/usernameHost Mask Port Status optipoint/optip 192.168.0.36(D) 255.255.255.255 1028 OK (118 ms) 19/19192.168.0.36(D) 255.255.255.255 1028 OK (131 ms) Now i can make calls to optipoint using 37 and make calls from optipoint using 19 in diferent contexts with includes. Note that * is registered like gateway in the optipoint conf pages regards Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP hackers gut Caller ID
This is very interesting... Regulations..USA... But... what can i do faking a caller id? stolen what? what is the point? miklos - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 12:56 PM Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID why regulate? nobody regulates the return address on a letter sent via USPS. - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 07, 2004 10:00 AM Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens Optipoint 400 SIP Problem
Hi! Yes we have many kinds of phones hwere in the show room, snom, polycom, cisco, grandstream, ipdialog, intracom, d-link, symbol all of them works with asterisk with some testing and with some issues ...but works. The optipoint is the only one that i´m really can´t make work till now. In the other side i can register the optipoint with FWD and Iconnecthere and use it really well The passwork field is really a problem...only acept numbers and more them 6 characters. I have port 5060 set up in the phone but it is registerisk with 1028. I will keep trying... regards Miklos ps. thanks to someone that alert me about the date in the mail, sorry about that list! - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Tuesday, June 22, 2004 6:26 AM Subject: Re: [Asterisk-Users] Siemens Optipoint 400 SIP Problem Thanks for your reply Miklos. I´m afraid I´m confronted with the same problem. Now my optipoint is registering to asterisk. (I had to configure the system type to SERVER , the Registrar Address Server Address to my asterisks-ip-address.) Now the optipoints are telling me: No Server... , that´s the same problem you have. Another problem is, that the opti can be called, but is not able to dial. Do you have a workaround for this? I tried different firmwares v2.16, v2.25, v2.3, v3 with TLS, but in all of them the same failure occured. Think I have to change to a different kind of hard-phone :-( Do you use another Hardphone ? Thanks in advance ! Roland Hi! I have updated the optipoint to the last software version I can Call the optipoint from other phones and talk. The optipoint register with asterisk but in the phone display i have only no server. and no dial tone. The only way to register was with no password to the optipoint peer. look at cli: -- Registered SIP 'optipoint' at 192.168.0.36 port 1028 expires 3600 -- Executing Dial(SIP/snom200-da4c, SIP/optipoint|20|tr) in new stack -- Called optipoint -- SIP/optipoint-a32b is ringing -- SIP/optipoint-a32b answered SIP/snom200-da4c -- Attempting native bridge of SIP/snom200-da4c and SIP/optipoint-a32b == Spawn extension (internal, 37, 1) exited non-zero on 'SIP/snom200-da4c' asterisk-eth0*CLI sip show peers Name/usernameHost Mask Port Status optipoint/optip 192.168.0.36(D) 255.255.255.255 1028 OK (130 ms) maybe that helps miklos - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 21, 2004 3:23 PM Subject: [Asterisk-Users] Siemens Optipoint 400 SIP Problem Hi there, I tried to get a few Optipoint 400 SIP working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I´m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I try to call one of the optipoints from a Sjphone, or from one opti to another, then no connection or signalling can be established. But Asterisk tells me that == Everyone is busy at this time. In my opinion, the optipoint is not registering correct to asterisk, when it is connected to the network.(By the way, SJphone does register to asterisk) So here is my question, does anyone suffer from the same problem and/or solved it??? Thanks a lot, greetz Roland (erlangen/germany) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Caller ID from FXO Problem
Hi! callerid=br exists? miklos - Original Message - From: Jason Williams [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 22, 2004 9:06 AM Subject: Re: [Asterisk-Users] No Caller ID from FXO Problem At 14:39 22/06/2004 +0300, you wrote: I've compiled and run it but no effect. Then i noticed that there is warning when i run asterisk Jun 22 14:37:47 WARNING[16384]: chan_zap.c:8762 setup_zap: Ignoring ukcallerid Make sure you have the correct switch in zapata.conf callerid=uk Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens Optipoint 400 SIP Problem
Hi! I have updated the optipoint to the last software version I can Call the optipoint from other phones and talk. The optipoint register with asterisk but in the phone display i have only no server. and no dial tone. The only way to register was with no password to the optipoint peer. look at cli: -- Registered SIP 'optipoint' at 192.168.0.36 port 1028 expires 3600 -- Executing Dial(SIP/snom200-da4c, SIP/optipoint|20|tr) in new stack -- Called optipoint -- SIP/optipoint-a32b is ringing -- SIP/optipoint-a32b answered SIP/snom200-da4c -- Attempting native bridge of SIP/snom200-da4c and SIP/optipoint-a32b == Spawn extension (internal, 37, 1) exited non-zero on 'SIP/snom200-da4c' asterisk-eth0*CLI sip show peers Name/usernameHost Mask Port Status optipoint/optip 192.168.0.36(D) 255.255.255.255 1028 OK (130 ms) maybe that helps miklos - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 21, 2004 3:23 PM Subject: [Asterisk-Users] Siemens Optipoint 400 SIP Problem Hi there, I tried to get a few Optipoint 400 SIP working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I´m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I try to call one of the optipoints from a Sjphone, or from one opti to another, then no connection or signalling can be established. But Asterisk tells me that == Everyone is busy at this time. In my opinion, the optipoint is not registering correct to asterisk, when it is connected to the network.(By the way, SJphone does register to asterisk) So here is my question, does anyone suffer from the same problem and/or solved it??? Thanks a lot, greetz Roland (erlangen/germany) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] embedded Asterisk
Hi! I will use it as simple ivr ...get the call from fxo gateway port ..give some options and rings the recepcionist phone. I have a x100p here and the thin client have a pci slot...maybe i can use it...maybe...not...i will test The main reason is to free a p4 2.0 ..that is runing * now... i think that it is to much only to say hello...press 1. :-) Miklos - Original Message - From: Stefan de Konink [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 16, 2004 8:22 PM Subject: Re: [Asterisk-Users] embedded Asterisk Probably the best thing to do is to build a uClibc tree, disable some Asterisk codecs (which don't want to compile, first run) compile again and run. Tomorrow I'm going to do the samething for an Epia-MII 1,2GHz/512MB/512MB-CF. Another tip :P Don't compile on flash... just make a tree on your harddisk. And copy the required binaries and libs to a root tree and attach a kernel. Look at some different Filesystems too, depending on for needs Ext2/Minix/CramFS. Btw. for what purpose do you want to run the box? I can imagine that a few voicemail messages can float the system. And if SIP is only required you should probably use SER for the project. I want to try out the VOCAL footprint too but didn't had the time to do that yet. Stefan listas iPfone wrote: Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip - NS Cx5530a Southbridge National Semiconductors SC2200 - NS PC97317 in chipset - 32MB Compact Flash - 64MB Ram - 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ideflash adaptor to make the install... I need recomendations in Linux distro... asterisk min. install ...etc..any info i can get. Thanks for any help Miklos Atenciosamente Cláudio Miklos * iP FONE *Telefonia IP Rua Caio Graco 735 São Paulo SP ( BR - 55 11 3801-3702 ( USA - 1 360-968-1591 ( FWD - 64662 ( sip:[EMAIL PROTECTED] www.ipfone.com.br http://www.ipfone.com.br [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] embedded Asterisk
Hi That rescue disk sugestion seems to be very good... Let´s see if i undestood: 1. burn the rescue iso 1. copy the rescue disk to a hard drive 2. compile asterisk 3. copy all to the flash disk It is that simple? Miklos - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 17, 2004 5:11 AM Subject: Re: [Asterisk-Users] embedded Asterisk Hi, Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you should be grand. Installing asterisk + some extra stuff will probably require, that you have at least a 128MB or 256MB flash or so. Dont go for stripped down but complete distributions which include a lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like i used the SuSE rescue system (14 mb), then you can add what you need (sshd,...) and compile asterisk on another box and then just copy it. My compressed ramdisk image is 32 mb, including all voice prompts and some mp3s for MOH. There are actually quite some board around on that CPU, like Soekris, pcengines and i think also Mikrotik at prices from 120EUR and up. I just put together the demo system for Linuxtag: - Via EPIA 5000 (C3-533), EUR 80,- - Morex case with external power supply, EUR 80,- - some old 256 mb SDRAMM - 128 MB USB memory stick, EUR 30,- - 1 quadBRI (could also easily handle an octoBRI, or a PRI card, with the dual riser pci card you can use 2 cards) The C3-533 is an i586 CPU. According to show translation it needs 30 ms for transcoding 1 channel from g711 to gsm (and vice versa). So, neglecting any overhead caused by channel handling it could transcode 30 channels to gsm. Linux BIOS has support for the EPIA boards, so you can speed up booting very much and also disable the VGA port (very useful for production deployments). I'm running pebble on a pcengines board, just needed to customize the kernel a bit, haven't been testing asterisk on that yet, but i definatly will in the sooner future. Kind regards, Martin List-Petersen martin (at) list (dash) petersen (dot) net best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] embedded Asterisk
Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip- NS Cx5530a Southbridge National Semiconductors SC2200- NS PC97317in chipset- 32MB Compact Flash - 64MB Ram- 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? Ihave a ideflash adaptor to make the install... I need recomendations in Linux distro... asterisk min. install ...etc..any info i can get. Thanks for any help Miklos Atenciosamente Cláudio Miklos iPFONE Telefonia IPRua Caio Graco 735 São Paulo SP ( BR - 55 11 3801-3702( USA - 1 360-968-1591 ( FWD - 64662( sip:[EMAIL PROTECTED] www.ipfone.com.br[EMAIL PROTECTED]
[Asterisk-Users] Asterisk Receptionist manager program asterisk girl 2004
Hi ! it was designed for our receptionist Please post a picture of that recepcionist .. maybe she can be the asterisk girl of 2004! Claudio - Original Message - From: Kyle Hagan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 03, 2004 6:21 PM Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. The Dial Pad is enabled in the newest executable only download. This is if you already have Asterisk Receptionist installed and just update the exe. Drop the exe in root:/ Program Files/Asterisk Receptionist or where ever you install it before. The reason it takes up the whole screen is it was designed for our receptionist, where it would always be up on the second monitor running 1024x768. I will look into making a smaller version. But this would allow less space for extention buttons. http://www.easyhomenetworks.com/AstRec/ Kyle Greg Blakely wrote: I had a similar result. The buttons work fine for transferring calls, but there was no dial pad shown. (Is there supposed to be?) Also, it would be VERY handy if it didn't have to take up the whole screen. I've taken to clicking on the icon in the upper left corner and choosing restore just so that I don't end up having to devote an entire workstation to nothing but Asterisk Receptionist. From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED] Sent: Thu 6/3/2004 1:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. It didn't work for me, didn't show me the keypad and my extention (IAX extention), below is a copy of the debug window. ---start--- Asterisk Call Manager/1.0 ---stop--- ---start--- Response: Success ---stop--- ---start--- Message: Authentication accepted ---stop--- ---start--- Response: Error ---stop--- ---start--- ActionID: 1 ---stop--- ---start--- Message: Permission denied ---stop--- --__--__-- Message: 2 Date: Thu, 03 Jun 2004 09:27:44 -0700 From: Kyle Hagan [EMAIL PROTECTED] Organization: Nuvo Technologies To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. Reply-To: [EMAIL PROTECTED] I put a new version up last night. Caller ID shows up on the buttons. This time IAX is fixed. Works at home and at work through FWD. http://www.easyhomenetworks.com/AstRec/ Has anyone had anyother bugs popup other than the IAX problem? Some people are asking why the screen shot has more buttons than the alpha version. We are going to get the bugs worked out of the existing buttons before we add more features. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INTERTEX AND ASTERISK
Hi all, I just upgrade my ix66 ... the new firmware 2.07 have this: (SIP) Tolerance against Asterisk PBX registration deviation. regards Miklos
Re: [Asterisk-Users] ATA devices
Audiocodes MP124 - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 18, 2004 12:45 PM Subject: [Asterisk-Users] ATA devices Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caller id detection
Hi! I know that is a very posted matter but i have a question: Some one can translate that messages for me? what is the mean of that messages? can i do something to correct this and get the caller id to work? May 7 11:26:19 ERROR[1288925632]: callerid.c:192 callerid_feed: fsk_serie made mylen 0 (-22)May 7 11:26:19 WARNING[1288925632]: chan_zap.c:4609 ss_thread: CallerID feed failed: SuccessMay 7 11:26:19 WARNING[1288925632]: chan_zap.c:4651 ss_thread: CallerID returned with error on channel 'Zap/1-1'May 7 11:26:19 NOTICE[1288925632]: chan_zap.c:3640 zt_read: Fax detected, but no fax extension Thanks fpr any help Miklos
Re: [Asterisk-Users] WI FI IP phones??
Symbol have the netvision line of h.323 wireless phones used in hospitals with multiple logins etc... , i have one here in my office and it works very well with a simple 3com officeconnect gateway, makes direct calls, have integration with various pbx.. a good product. www.symbol.com Miklos - Original Message - From: Mark Musone [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 2:20 PM Subject: RE: [Asterisk-Users] WI FI IP phones?? Why not vocera? http://www.vocera.com they seem to have the exact product you are looking for and seem to primarily server hospitals.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Moran Sent: Friday, May 07, 2004 1:06 PM To: Asterisk Subject: Re: [Asterisk-Users] WI FI IP phones?? Hmm I'll look into it. Thanks. On Fri, 2004-05-07 at 12:54, John Fraizer wrote: James Moran wrote: No I'm not but it's a hospital that nurses are on call and need to have a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote: James Moran wrote: We need to have about 30 phones on one floor And you really think that WiFi phones are suited for this application? Not an RF engineer, are ya? John Um, I'm not so sure that you're going to be able to run WiFi at a hospital. The life safety/support equipment is most likely not certified to be resistant to 2.4Ghz interference. It's been a while since I looked up ISM allocations but, I can tell you that I've seen many good ideas shot down because of the potential to interfere with the medical equipment. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Moran [EMAIL PROTECTED] Potential Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Wokflow diagram
SIP Scenario Generator http://www.ipc.com/ runs under windows Miklos - Original Message - From: Brancaleoni Matteo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 07, 2004 2:51 PM Subject: Re: [Asterisk-Users] SIP Wokflow diagram I use callflow (callflow.sourceforge.net) works under linux with ethereal dump, and produces html+images pages, for viewing them via a web browser. Matteo. Il ven, 2004-05-07 alle 15:14, Ignace CARIA ha scritto: Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h.323 show codecs was WARNING[1074420448]
Thanks Jeremy, The problem is ended now. But... when i use de h.323 show codecs nothing happens... my h323.conf have the lines: disallow=all allow=all ; turns on all installed codecs ;disallow=g723.1 ; Hm... Proprietary, don't use it... ;disallow=all ; Disallow all codecs ;allow=ulaw ;allow=alaw ;allow=gsm ;allow=g729 ;allow=ilbc I´m doing something wrong here?? miklos - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, April 23, 2004 7:18 PM Subject: Re: [Asterisk-Users] WARNING[1074420448] listas iPfone wrote: plase somebody help me... You should help yourself first and read the asterisk/channels/h323/README. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] siemens optipoint 400 sip
Hi list I have configured some siemens optipoint 400 sip to work with asterisk. I works very well with messages, moh etc... a good choice in my opinion... Someone else have good/ bad experiences with that phones? Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Noises and echo effects
Olá Ana, Estou aguardando as informações sobre nosso acordo de revenda Atenciosamente Cláudio - Original Message - From: Adam Goryachev [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 31, 2004 5:25 AM Subject: Re: [Asterisk-Users] Noises and echo effects Doesn't help much, but I have the same problem. Also same problem to some normal land lines (I suspect it is to any other digital service, so calls to other E1 services have the same problem). To make you feel better, I know other people have this setup and it works correctly, I just haven't really had a chance to look into it... I am using a TE405P - telco E1 service, no additional equipment involved. I notice it mostly when calls arrive on the E1 and forward (on another channel of the same E1) to my mobile. Users report the same when calling from a TDM40B to mobiles/digital landlines using the E1. It also seems to happen to calls arriving on the X100P and outbound on the E1 to mobiles. PS, my callerid doesn't work on the TE405P either, but that is even lower priority ... Regards, Adam On Wed, 2004-03-31 at 18:03, Serge Oleinikov wrote: Hi! I need your advice. My problem is that I have very bad sound quality calling to cellular phone via asterisk router. There are some kind of noises and echo effects when you try to speak louder. -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax detected, but no fax extension
Hi! Every time i make or receive a call with my x100p i receive that notice: NOTICE[1234379840]: chan_zap.c:3640 zt_read: Fax detected, but no fax extension Maybe that is problem with brazilian lines? How can i stop it? Miklos iPFONE Telefonia IPRua Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702FWD 64662sip:[EMAIL PROTECTED] www.ipfone.com.br[EMAIL PROTECTED]
[Asterisk-Users] Callerid detection
Hi All! I have this problem with callerid detection with my x100p here in brazil., my line have this function and it works with a very cheap aplliance that i have here in the office, here in brazil it is called "detecta". Ithink that the caller id info comes in DTMF before the 2 ring of the incoming call, so i think that because asterisk is answering the call in the 1 ring it can´t identify the callerid info. There is a way to make asterisk wait for the second ring to see ifit identifies the callerid info? I don´t know if myidea is correct, anyone have some sugestion on how to make asterisk identify thecallerid here in brazil? Thanks for all Miklos
Re: [Asterisk-Users] Callerid detection
Ok! I hope some *guru can make it soon... :-) but i´m happy to know that my guess is correct! thank´s Miklos - Original Message - From: Alfred R. Nurnberger To: [EMAIL PROTECTED] Sent: Tuesday, February 10, 2004 12:48 PM Subject: RE: [Asterisk-Users] Callerid detection You are right, Brazil uses DTMF caller ID. The format is very simple Dtmf-DNUMBERDtmf-C Asterisk has all the tools available to get DTMF caller ID to work. (DTMF decoder routines,etc.)and T1-CAS uses a very similar format. I guess somebody just needs to spend the time and programm it into the zaptel driver. Alfred. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of listas iPfoneSent: Tuesday, February 10, 2004 8:20 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Callerid detection Hi All! I have this problem with callerid detection with my x100p here in brazil., my line have this function and it works with a very cheap aplliance that i have here in the office, here in brazil it is called "detecta". Ithink that the caller id info comes in DTMF before the 2 ring of the incoming call, so i think that because asterisk is answering the call in the 1 ring it can´t identify the callerid info. There is a way to make asterisk wait for the second ring to see ifit identifies the callerid info? I don´t know if myidea is correct, anyone have some sugestion on how to make asterisk identify thecallerid here in brazil? Thanks for all Miklos
[Asterisk-Users] compact fxo device
Hi All! I´msearching fora compact external fxo device , a little box like sipura adaptor,with one or maybe two fxo. Searching google the only device that shows is the x100p, Anyone knows about a device like that? miklos
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
Snom Does gives the souce and more: http://www.snom.com/sources_en.php - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 4:01 PM Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junk calls from FWD numbers
Hi! If the number of calls are really greate maybe you are listed in the fwd welcome (5) line by mistake... Miklos - Original Message - From: Andy Powell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 29, 2004 9:53 AM Subject: Re: [Asterisk-Users] Junk calls from FWD numbers *** REPLY SEPARATOR *** On 27/01/2004 at 15:55 Chris Albertson wrote: My Asterisk server registers two FWD numbers. On average I get about one call a day from someone calling from an FWD number and leaving a pointless, under 10 second message. It's easy to see who these people are if I look in my CDR file I can see thier name and number. They seem to be new FWD users, likely who've just downloaded FWD's Xten softphone and then dial some random FWD user (me) to try it out. I wonder if these same people when they first got a POTS phone installed in thier home got out the white pages and dialed randomly asking anyone who'd answer Hi does this work? can you hear me? Question: Does everyone with an FWD number get these junk calls or am I the only lucky one? There are a number of things you can do: 1. Make sure you are not listed in the white pages (turn it off from your settings page) since FWD is a community it's pretty much accepted that if you list in the white pages you are open to receiving calls from people you don't know. Hopefully they are at least respecting your timezone settings. The FWD white pages bears no resemblance to a 'normal' white pages .. they share only a name. 2. If the calls are nusance calls then get in touch with Ed Guy and report the problem - (Don't rely on the caller id as to where the call came from.) 3. Keep in mind the reason that you don;t get calls on your pstn line with people saying Hi does this work? can you hear me? is because pstn calls are tried and tested over many years. voip doesn't have this (pstn calls don't get NAT issues) luxury. There is of course nothing you can do about people dialing random digits... HTH Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rc.local dont works
Hi Jeroen1 I think that´s maybe a bug I really don´t found the problem in my logs, i´m starting it by hand :-( I update you if i can figure it out. regards Miklos - Original Message - From: Jeroen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 26, 2004 11:23 AM Subject: Re: [Asterisk-Users] rc.local dont works Hi Miklos, I have the same problem here in RH90 - have you found any solution? Or does anybody else know why (safe_)asterisk does not start using rc.local? (normally I start * as root user) Cheers Jeroen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rc.local dont works
Ok! Thanks miklos - Original Message - From: Karsten Wemheuer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 24, 2004 9:42 AM Subject: Re: [Asterisk-Users] rc.local dont works Hi Miklos, listas iPfone wrote: Hi ! thanks for the answer.. I use rh9... Sorry, I am familiar with Linux From Scratch, Debian and Gentoo but not with RH. I think with an interrupt problem, any startup will fail, may it be manual or automatic during startup. but.. you think that there is a problem in the interrupts at all? i don´t understand. Sorry, that was a little bit irritating from me. English is not my mother tongue. I mean, if it is an interrupt problem, there would be no difference in the results. So, I think it is NOT an interrupt problem. HTH HAND Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Absolute Timeout
I use it in that way, it works very well: exten = s,4,AbsoluteTimeout,600 miklos - Original Message - From: Wes Marderness [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 12:33 PM Subject: [Asterisk-Users] SIP Absolute Timeout Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten = _X.,1,Absolutetimeout(20) exten = _X.,2,dial(SIP/[EMAIL PROTECTED]) exten = T,1,BackGround(tt-weasels) exten = T,2,Hangup() Thanks ahead of time for any help / suggestions. Wes Marderness ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 phones not working.
Hi I sugest you to make a reset and switch off the phone before upgrade. It solved many problems for me. Miklos - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 11:32 AM Subject: Re: [Asterisk-Users] Snom 200 phones not working. Ariel, I have 2 Snom 200 and would like to get them to work properly with Asterisk. With the Firmware 2.02t I am able to use the phone. But only one line configured. With there newer firmware 2.03o it will not allow me to make calls. But I can get calls on the unit. Again the 2nd line is not able to be registered. Is this an issue with Asterisk or Snom? I could use some example configuration files. I have followed the Snom FAQ step by step. But it's still not working. I just upgraded my 200 to v2.03o and its working fine with two extns defined. I happen to be using * CVS-12/04/03-14:24:40 on the same wire (no nat, etc). My sip.conf entries look like: [3007] type=friend host=dynamic username=3007 secret=mypassword context=from-sip [3008] type=friend host=dynamic username=3008 secret=mypassword context=from-sip Using your web browser to config the phone, verify: Settings/SIP/Lines Account = 3007 (to match above sip.conf def) Registrar = ip address of asterisk box Action = proxy Account = 3008 (to match above sip.conf def) Registrar = ip address of asterisk box Action = proxy Settings/SIP/Stack Outbound Proxy, Registrar is outbound proxy = yes Settings/SIP/Authentication Line 1, Realm = asterisk, Username = 3007, Pasword = mypassword Line 2, Realm = asterisk, Username = 3008, Pasword = mypassword Settings/Key Mapping P1 = Line, Number = 3007 P2 = Line, Number = 3008 After ensuring your phone settings actually match the sip.conf settings and that you've properly selected Save after changing each of the above entries in the phone, then reboot the phone. If the phone prompts you to download another firmware image, simply press ESC. (Seems some config changes don't take effect until after a phone reboot.) The above config has been working fine with the last several (estimate about 10) firmware versions, however the user interaction with several of the keys are rather non-intuitive (or even backwards) for US users. For example, if you answer an incoming call on Line 1 (x3007 above) and place that call on hold using the Hold key, then select Line 2 (x3008) to do a consultive call to a different extn, you have to press the ESC key to hang up that second consultive call. If instead of pressing the ESC key you simply press Line 1 to return to the original call, Line 2 is automatically put on hold (instead of dropping the line as it does in the US). If you're not paying attention to the LEDs, you've now tied up the second line/extn until such time as you muck around to release it. If that second (consultive) call happens to be to a pstn user and your Central Office supports calling-party line supervision, you've probably tied up that person's telephone line as well. (Email comments to snom resulted in push-back, suggesting the ESC key is the proper way to drop that second line. I'd guess US users (not techie's) will object to using the phone in any form of production telephony.) I've not tried the 200 with the later CVS versions, so don't have a clue as to what you're milage may be. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rc.local dont works
Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don´t in the initialization... I have in my file that comands: touch /var/lock/subsys/localmodprobe zaptelmodprobe wcfxosafe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody can tell me if this is correct? The usb-ohci and usb-ohci drivers are to be sharing the same interrupt as the wcfxo? CPU0 0: 220155 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 68 XT-PIC eth1 8: 1 XT-PIC rtc 9: 2167768 XT-PIC usb-ohci, usb-ohci, wcfxo10: 7320 XT-PIC eth012: 22 XT-PIC PS/2 Mouse14: 5092 XT-PIC ide0NMI: 0ERR: 0 Thanks for any help Miklos
Re: [Asterisk-Users] rc.local dont works
Hi ! thanks for the answer.. I use rh9... I think with an interrupt problem, any startup will fail, may it be manual or automatic during startup. but.. you think that there is a problem in the interrupts at all? i don´t understand. regards Miklos - Original Message - From: Karsten Wemheuer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 5:07 PM Subject: Re: [Asterisk-Users] rc.local dont works Hi listas iPfone wrote: Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don´t in the initialization... If it works when called directly and it doesn't work called during startup. I would think, it is a problem with path-setting or with access rights. The init-scripts normaly have a relative short path. Maybee the executable is not found. Or Your setup starts * as a non-root user, but your manual startup uses root. What distribution do You use? I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening I think with an interrupt problem, any startup will fail, may it be manual or automatic during startup. HTH, Karsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and rh9 boot problem
Hi All! I installed * in RH9 with yesterday cvs and i have a x100p in that system. My problem is that when rh9 loads, it loads the zaptel modules ( wcfxo and the usb driver) automagically, and when it calls my rc.local with: modprobe zaptelmodprobe wcfxosafe_asterisk asterisk dont start. I don´t need the usb module because i only have the x100p in the system... anyone knows why it loads in the boot? and how can i stop it? In the previuos version with RH8 it only loads with the rc.local...i´m confuse. thanks Miklos
[Asterisk-Users] ERROR[8192]
Hi all! I get this error when trying tostart asterisk: ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk What can be the problem? Thank you! Miklos iPFONE Telefonia IPRua Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702UK +44 870 - 3403539FWD 64662sip:[EMAIL PROTECTED] www.ipfone.com.br[EMAIL PROTECTED]
[Asterisk-Users] ultra-cheap (and easy) asterisk box
I think that it will be greate to include * inside of a router like ix66 from intertex... 1 GB usb removable flash to record voice mail.and prompts in the computer..2 fxo...real internal sip server ...internal dns server..good user interface.. all nat / firewall nightmare ended, no computers to worry about. Just dreaming with my little office pbx for about $200 regards Miklos - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 3:31 PM Subject: Re: [Asterisk-Users] ultra-cheap asterisk box I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. Ideally Asterisk will be ported one day to Linux/ARM or some other very low cost platform. for VOIP you do not need the PCI slots. In theory Asterisk could run on a Lynksys router box with re-flashed EEPROM. After all Lynksys' latest wireless router runs Linux inside Low cost to me means low total cost of ownership To get this I don't think buying the lowest priced parts is the way to go. I want quality mainboard, and a quality power supply and, this is importernt: A low internal case temperature. for this reason I'll spend the extra $50 to go with Antec cases and ASUS mainboards over the generic ones. What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Symbol NetVision Phone
HiList ! I received an unit of the Symbol NetVision Phone and i will test it with asteriskusing H.323 or Skinny , somebody tested thisphone with asterisk and can share experience? Miklos
Re: [Asterisk-Users] 128 kbs satelite link
Hi! Last week i talk to a person in senegal (i´m in brazil) with a 64 Kbs sattelite link and the latency was about 10 seconds! Like you are talking to the moon. miklos - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 17, 2003 1:52 PM Subject: RE: [Asterisk-Users] 128 kbs satelite link David Gomillion wrote: Senad Jordanovic wrote: Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? One word of caution: you may have latency problems. Even at the speed of light, the information has a LONG way to travel... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Sure, valid point you are making! However, devices like Cisco 53XX (and others) do put 9 calls through 128 kbs using G723. I have not tested it my self , but the www.transcom.com people say it can be done. :) I presume if G729 where to be used instead then 6 calls would be the figure. Now the question is: Can IAX be a BIG help here due to its method of sending packets? Also is IAX/IAX2 suitable for C or KU band satelites? BTW, does anyone know another provided similar to www.transcom.com ? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mysql CDR
Thanks for all! It is working now :-) Regards Miklos - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, December 13, 2003 3:16 PM Subject: Re: [Asterisk-Users] Mysql CDR On Saturday 13 December 2003 11:02, Mireia Munoz de jesus wrote: The line with ;sock=/tmp/mysql.sock, i think you must write it without the ;. You need this socket to connect with mysql. You don't usually need that configuration line. It's only there if your server and client conflict about the correct location for the sock file. For example, the MySQL default location is /tmp/mysql.sock, but RedHat (and others) insist on putting that in /var/lib/mysql/mysql.sock. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 200 version 2.03b with changed music on hold
Hi! i just tryied the 2.03b firmware. Now i have that message when the phone boots: Challenge User: 6466212364662 64662 pressing ok the display shows PW: iputmypassword When i put my password i get a loop returning for Challenge User: 6466212364662 again 64662 is my FWD number Now the phone don´t register with fwd anymore. And more... I have two snom phones.. one 100(firmware 1.16x) and one 200( with this new firmware). When i call the snom 100 from the snom200 and put the call on hold i have moh from asterisk in the snom100, vice versa don´t works...no moh at all. i hope that helps. regards Miklos - Original Message - From: Christian Stredicke [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 15, 2003 2:23 PM Subject: [Asterisk-Users] snom 200 version 2.03b with changed music on hold Hi folks, in order to establish backward compatibility we made an image that automatically detects if the other side does not support RFC3264. Please try it out, we would be very interested if this image is a progress! http://snom.com/download/share/snom200-2.03b-SIP.bin Thanks, CS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT How to do it.
Hi The version 1.260 of chan_sip.c already have that patch?: http://bugs.digium.com/file_download.php?file_id=430type=bug thanks! Miklos - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 28, 2003 2:10 AM Subject: [Asterisk-Users] Asterisk behind NAT How to do it. Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.cStatus: Locally Modified Working revision:1.258 Repository revision: 1.258 /usr/cvsroot/asterisk/channels/chan_sip.c,v While in pwd /usr/src/asterisk/channels/ patch -p0 /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom X MOH
Hi! Only to make clear... As a brazilian..i love the "bossa nova" MOH from snom and the sound quality is very good. Yesterday it was playing "barquinho" from João Gilberto :-)) regards Miklos - Original Message - From: Christian Stredicke To: [EMAIL PROTECTED] Cc: 'Robert Messer' ; 'Kevin' ; [EMAIL PROTECTED] ; [EMAIL PROTECTED] ; [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 7:22 AM Subject: AW: [Asterisk-Users] snom X MOH Dear All, it seems there is some confusion about using MOH correctly. RFC3264 (http://ietf.org/rfc/rfc3264.txt?number=3264) describes the usage of the a=recvonly and sendonly parameters of SDP. Using the address 0.0.0.0 is depreciated. We have moved to be RFC3264 compatible and seems to cause the problem. It seems other vendors did not do this step yet; however I assume that they will follow sooner or later. Now what can we do? Option A would be moving snom backwards and use 0.0.0.0. Option B would be to move Asterisk forward and handle the a=send tags. As I am generally against moving backward, I think its better to include RFC3264 handling in Asterisk. Until then, using snom2.02t together with the current Asterisk will result in missing music on hold. If there should be any bug on our side using RFC3264 correctly, we will surely fix this as soon as possible. Best, Christian -Ursprüngliche Nachricht-Von: Kevin [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 9. Dezember 2003 04:50An: [EMAIL PROTECTED]Cc: 'Robert Messer'Betreff: FW: [Asterisk-Users] snom X MOH Hi CS, Hope all is well. Did mr. Spence get back to you on this? I think I sent you all the info you needed. It appears other asterisk users are hitting this problem on 2.0 and dont like it!! Can you check it out? Thanks, Kevin -Original Message-From: Ernest W. Lessenger [mailto:[EMAIL PROTECTED] Sent: Monday, December 08, 2003 3:44 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] snom X MOH At 12:23 PM 12/8/2003, "listas iPfone" [EMAIL PROTECTED] wrote: I updated my snom200 to 2.02t and now MOH from * don´t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension).Someone with that problem? I am having the same problem. You can resolve it temporarily by downgrading to the 1.6.x series of SNOM. I am BCC'ing this email to a SNOM representative who is working on this issue.--Ernest
Re: [Asterisk-Users] IpDialog phone issues.
Hi! I have one ipdialog working well with cvs 10/09 but with latest cvs i have the same problem. regards miklos - Original Message - From: Ariel Batista [EMAIL PROTECTED] To: Asterisk User List [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 5:34 PM Subject: [Asterisk-Users] IpDialog phone issues. I have gotten this phone to work with SIP configuration. It really sounds great. The only problem is with voicemail2. When I access voicemail the voicemail2 will not respond to the digits. I can transfer to any extension and Asterisk picks up the digits then. But once in the Voicemail2 program it fails? Any Ideas? Here is my configuration in the sip.conf. [ipdialog2] type=friend context=main username=ipdialog2 secret=X host=dynamic dtmfmode=rfc2833 ; I have tried inband and info. nat=1 disallow=all allow=ulaw allow=alaw [EMAIL PROTECTED] Ph: 786-544-1114 fwd:700-544-1100x114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom X MOH
Hi all! I updated my snom200 to 2.02t and now MOH from * don´t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have noMOH at all..( with the transfer button, moh playsusing a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos
Re: [Asterisk-Users] Asterisk behind NAT How to do it.
Hi! I need help to undestand the options: externip= static/ dynamic ip? can be a domain? localnet= internal ip of * machine? localmask= 255.255.255.0 ? Thanks - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 7:25 AM Subject: RE: [Asterisk-Users] Asterisk behind NAT How to do it. On Tue, 2003-12-02 at 15:55, Arnold Ligtvoet wrote: Hi Leif, I tried the patch. Installed it exactly as described per your email. Thought that you might be interested that it works for me as well. Like a charm, I can finally call FWD numbers like 10001 and 612 (speaking clock demo). BTW: For anybody wanting to install this, if your version of chan_sip.c is older than the one described, first use 'cvs update -C asterisk/channels/chan_sip.c'. Awesome! Have you tried the newer patch / diff for 1.259 (which as of right now is the newest chan_sip file). If you goto bugs.digium.com and login anonymously and jump to bug 104, then you can get the newest patch. Same instructions as before. I just updated it to test the new sip.conf structure which is externip= localnet= localmask= Still working great for me here! BTW! Can you login to the bug tracker and post a comment ? Thanks! -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] test call request
Hi all! We set up a sipserver using asteriskX ix66 and need some test calls from around world toverify if it is working ok. If you can :-)please call us: sip:[EMAIL PROTECTED] direct to snom200 or sip:[EMAIL PROTECTED] to asterisk snom200 Thank´s for all Miklos iPFONE Telefonia IPRua Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702FWD 64662ICH 31451543www.ipfone.com.br[EMAIL PROTECTED]
Re: [Asterisk-Users] test call request
Hi !
Thank you for the call
I think that you have to Put reinvite=no in your sip.conf for the given
friend/user/peer to keep * from trying a native bridge.
I tryed to call you ( sip:[EMAIL PROTECTED] and
sip:[EMAIL PROTECTED]) but the call timeout
Thank you again
Miklos
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 4:02 PM
Subject: Re: [Asterisk-Users] test call request
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls
from around world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] direct to snom200
or
sip:[EMAIL PROTECTED] to asterisk snom200
Thank?s for all
Miklos
Miklos,
OK, I just dialed, looks like you answered. However my * attempts a
native bridge between my grandstream phone and your sipserver. Do you have
a suggestions on how I can set up a stanza in sip.conf so I can call you and
keep * from trying a native bridge?
--console log:
-- Executing Dial(SIP/2400-3989, sip/[EMAIL PROTECTED]|60)
in new stack
-- Called [EMAIL PROTECTED]
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 answered SIP/2400-3989
-- Attempting native bridge of SIP/2400-3989 and
SIP/sipserver.com.br-c906
--extensions.conf:
exten = 90,1,Dial(sip/[EMAIL PROTECTED]|60)
exten = 90,2,Hangup
Thanks, Walker
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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Re: [Asterisk-Users] test call request
Hi dave
I think that is a problem with nat, calls direct to the snom phone trough
ix66 works well but from asterisk don´t.
Thanks for the call
Miklos
- Original Message -
From: David J Carter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 24, 2003 5:04 PM
Subject: RE: [Asterisk-Users] test call request
Hi Miklos,
I have the same as Walker.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Walker Haddock
Sent: 24 November 2003 18:02
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] test call request
On Mon, Nov 24, 2003 at 02:10:48PM -0200, listas iPfone wrote:
Hi all!
We set up a sipserver using asterisk X ix66 and need some test calls
from
around world to verify if it is working ok.
If you can :-) please call us:
sip:[EMAIL PROTECTED] direct to snom200
or
sip:[EMAIL PROTECTED] to asterisk snom200
Thank?s for all
Miklos
Miklos,
OK, I just dialed, looks like you answered. However my * attempts a
native
bridge between my grandstream phone and your sipserver. Do you have a
suggestions on how I can set up a stanza in sip.conf so I can call you and
keep * from trying a native bridge?
--console log:
-- Executing Dial(SIP/2400-3989, sip/[EMAIL PROTECTED]|60)
in
new stack
-- Called [EMAIL PROTECTED]
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 is ringing
-- SIP/sipserver.com.br-c906 answered SIP/2400-3989
-- Attempting native bridge of SIP/2400-3989 and
SIP/sipserver.com.br-c906
--extensions.conf:
exten = 90,1,Dial(sip/[EMAIL PROTECTED]|60)
exten = 90,2,Hangup
Thanks, Walker
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
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[Asterisk-Users] help voicepulse connect
Hi All I signed up for an account with voicepulse connect service and received the info to set up asterisk. Anyonehave that confs to send as an example? Thanks Miklos
Re: [Asterisk-Users] Open Source Linux PBX!
Title: Mensaje Try this guide: http://www.automated.it/guidetoasterisk.htm Miklos - Original Message - From: Sergio Serrano Revuelto To: [EMAIL PROTECTED] ; [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Thursday, November 13, 2003 8:02 AM Subject: RE: [Asterisk-Users] Open Source Linux PBX! try to cvs srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Quan Le TrungEnviado el: jueves, 13 de noviembre de 2003 10:43Para: [EMAIL PROTECTED]CC: [EMAIL PROTECTED]; [EMAIL PROTECTED]Asunto: [Asterisk-Users] Open Source Linux PBX! Hi! I have just bought the Wildcard TDM400P 4-port FXS PCI Card, and Wildcard X100P is a single-port FXO PCI Card to install on my computer to implement the PBX (Private Packet Exchange). However, I cannot download the corresponding softwares (asterisk, libpri and zaptel) at the following address: ftp://ftp.asterisk.org/pub/telephony . If anyone has already downloaded these softwares, please kindly send them to me via the following e-mail: [EMAIL PROTECTED] . Thanks in advance! P.S Please kindly send files in separate e-mails to me because of limited size of received e-mails. Best regards, Quan L. T.
[Asterisk-Users] INTRACOM SIP PHONE
Hi all! I´m testing an intracom sw netphone with asterisk, someone have one netphone or have any experience to share about? miklos
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Hi! How to use that externip new parameter? Where in sip.conf and what is the format? thanks - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 03, 2003 3:34 PM Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Martin On Mon, 3 Nov 2003, WipeOut wrote: Martin Pycko wrote: You can port forward the 5060 SIP port and use externip keyword in sip.conf to have it working behind a NAT. Martin Martin, Is externip and new parameter?? Does it do a similar thing for the server as what nat=yes does for the phone? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on remote end when using NuFone
I have the same problem and it was solved setting: # Uncomment for aggressive residual echo supression under # MARK2 echo canceller # KFLAGS+=-DAGGRESSIVE_SUPPRESSOR in the makefile of zaptel and recompiling. miklos - Original Message - From: Ernest W. Lessenger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 31, 2003 4:21 PM Subject: [Asterisk-Users] Echo on remote end when using NuFone I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a separate issue, I am hearing a bad echo when using my Digium X100P to connect to the PSTN. I've tried tweaking the tx/rx gain to no real effect. I've also tried changing the volume on the SNOM phone, changing the codec to g711u, and decreasing the packet size. Any other things to try? Thanks, --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnecthere connect problem
Hi!
try to use in sip.conf :
register =x:[EMAIL PROTECTED]/xx
[iconnect]
type=friend
secret=
username=xxx
host=sipauth.deltathree.com
dtmfmode=inband
context=yourcontext
and in extensions.conf:
exten = _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
This works for me
regards
Miklos
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 25, 2003 5:17 PM
Subject: [Asterisk-Users] Iconnecthere connect problem
I have an Asterisk box behind NAT and am trying to connect to Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial(SIP/2001-12c8, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
== No one is available to answer at this time
sip.conf is:
[delta3]
type=peer
username=
secret=
host=213.137.73.140
the extension.conf entry is:
exten =_1706NXX,1,Dial,SIP/[EMAIL PROTECTED]
Am I missing something??
Robert
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Re: [Asterisk-Users] Iconnecthere connect problem
Hi!
I don´t have an inbound number to, this registration is for an outbound
account
sorry if i don´t explain better in he first time
register=username:[EMAIL PROTECTED]/extension
hope this helps
miklos
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 8:49 AM
Subject: Re: [Asterisk-Users] Iconnecthere connect problem
Hello..
Thanks for the reply.. I'll give this a check later today. Is the first
x in the register command your phone number at ICONNECTHERE? I am
using them with the demo account only as outbound so don't have a phone
number. Maybe this could be the problem.
Regards,
Robert
Friedriedrichshafen, Germany
Hi!
try to use in sip.conf :
register =x:[EMAIL PROTECTED]/xx
[iconnect]
type=friend
secret=
username=xxx
host=sipauth.deltathree.com
dtmfmode=inband
context=yourcontext
and in extensions.conf:
exten = _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
This works for me
regards
Miklos
- Original Message -
From: rnc Info Lists [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 25, 2003 5:17 PM
Subject: [Asterisk-Users] Iconnecthere connect problem
I have an Asterisk box behind NAT and am trying to connect to
Iconnecthere
as was indicated possible earlier. Am getting the following on the
Asterisk console:
-- Executing Dial(SIP/2001-12c8, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
== No one is available to answer at this time
sip.conf is:
[delta3]
type=peer
username=
secret=
host=213.137.73.140
the extension.conf entry is:
exten =_1706NXX,1,Dial,SIP/[EMAIL PROTECTED]
Am I missing something??
Robert
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[Asterisk-Users] how to use gastman/astman?
Hi! where i can find info about using gastman and astman? Thanks! Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING[49159]
Hi All I receive thatwarning message: WARNING[49159]: File chan_sip.c, Line 2220 (__transmit_response): Unable to determine sequence number from '' What is it? There is some documentation with all error messages? thanks miklos
Re: [Asterisk-Users] WARNING[49159]
];tag=483a-f0f0b8ca To: 35 sip:[EMAIL PROTECTED];tag=as3028bf6d Call-ID: [EMAIL PROTECTED] CSeq: 26289 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 180 Contact: sip:[EMAIL PROTECTED];expires=180 Date: Tue, 14 Oct 2003 16:30:06 GMT Content-Length: 0 to 192.168.0.33:5060 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK15438552 From: asterisk sip:[EMAIL PROTECTED];tag=as787ccf10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/1 (no NAT) to 192.168.0.33:5060 Sip read: CLI SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.31:5060;branch=z9hG4bK15438552 Call-ID: [EMAIL PROTECTED] Contact: 35 sip:[EMAIL PROTECTED] CSeq: 102 NOTIFY From: asterisk sip:[EMAIL PROTECTED];tag=as787ccf10 Supported: timer To: sip:[EMAIL PROTECTED];tag=02f8-f0f0f208 Server: ipDialog SipTone 1.2.0 rc V UAS Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY Content-Length: 0 11 headers, 0 lines localhost*CLI - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 12:39 PM Subject: Re: [Asterisk-Users] WARNING[49159] It means that your SIP device sends some SIP packets and we can't parse the CSeq numbers. Can you paste the 'sip debug' of that ? regards Martin On Tue, 14 Oct 2003, listas iPfone wrote: Hi All I receive that warning message: WARNING[49159]: File chan_sip.c, Line 2220 (__transmit_response): Unable to dete rmine sequence number from '' What is it? There is some documentation with all error messages? thanks miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] my phone shows asterisk
Hi! My setup is: pstn X100PASTERISKSNOM 200 thanks miklos - Original Message - From: Gerry Boudreaux [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 09, 2003 8:12 PM Subject: Re: [Asterisk-Users] my phone shows asterisk What hardware are you using to connect to the PSTN? G At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote: Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] my phone shows asterisk
Hi! Thanks for the advice i will do it. There is a way to know if the CallerID enabled from my telco is compatible with asterisk? regards Miklos - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 8:08 AM Subject: Re: [Asterisk-Users] my phone shows asterisk listas iPfone wrote: Hi! My setup is: pstn X100PASTERISKSNOM 200 thanks miklos - Original Message - From: Gerry Boudreaux [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 09, 2003 8:12 PM Subject: Re: [Asterisk-Users] my phone shows asterisk What hardware are you using to connect to the PSTN? G At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote: Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos I am guessing your don't have CallerID enabled from your telco.. If you do it is probably incompatible.. If you really hate the asterisk showing on your screen then use the SetCallerID and SedCIDName commands on your inbound calls.. eg.. exten = s,1,SetCallerID(555 4321) exten = s,2,SetCIDName(Inbound Call) exten = This will obviously be statically set and will not show the CallerID of the person that is calling but it wil get rid of the word asterisk on your screen.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] telefonica sp brazil caller id problem
Hi I have problems with caller id in my line from telefonica sp brazil, anyone knows if there is any problem with this telco caller id and asterisk? thanks miklos - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 10:10 AM Subject: Re: [Asterisk-Users] my phone shows asterisk listas iPfone wrote: Hi! Thanks for the advice i will do it. There is a way to know if the CallerID enabled from my telco is compatible with asterisk? regards Miklos I guess if it is enabled and it does not work then chances are that it is not going to be compatible.. Post who your Telco is and maybe there is someone else who knows.. later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] my phone shows asterisk
Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] runing asterisk and apache
Hi All, I´m thinking in install apache in my asterisk machine to host a litle site. Anybody knows about problems doing that? thanks miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs questions
Hi! I have some question about the use of codecs in sip.conf I have that lines in sip.conf: disallow=all allow=gsm allow=ulaw allow=alaw when i use show codecs: localhost*CLI show codecs 1 (1 0) G.723.1 2 (1 1) GSM 4 (1 2) G.711 u-law 8 (1 3) G.711 A-law 16 (1 4) MPEG-2 layer 3 32 (1 5) ADPCM 64 (1 6) 16 bit Signed Linear PCM 128 (1 7) LPC10 256 (1 8) G.729A audio 512 (1 9) SpeeX 1024 (1 10) iLBC 65536 (1 16) JPEG image 131072 (1 17) PNG image 262144 (1 18) H.261 Video 524288 (1 19) H.263 Video My questions are: 1) What is the best configuration to use with fwd? 2) my sip.conf is correct? I can make calls to fwd but i have problems to listen to people that calls me. 3)Asterisk is using the G.723.1 as the first choice? 4) I have to configure my phones with the same codec that asterisk is using or the interoperable option in the snom phone is correct? Thanks Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect Incomming calls
Hi!
I´m thinking inan incoming number from
ICH
please share your sip and extensions.conf files off
list, it will help me a lot.
miklos
- Original Message -
From:
Glenn
Dalgliesh
To: [EMAIL PROTECTED]
Sent: Friday, October 03, 2003 2:17
PM
Subject: [Asterisk-Users] Iconnect
Incomming calls
I have an IconnectHere
account with a Inbound number and have setup the sip.conf to register and am
recieving the call but When I answer the call it disconnect. I have tried
sending the call to from * to a Softphone, Pingtel, and FXS port and all
result the same. As soon as I accept the call it disconnects. I believe it may
be some type of codec issue but I am not very familiar with that
layer.
Below is the SIP
debug
Thank for any
help
to
162.33.165.195:5060Sip read: INVITE sip:[EMAIL PROTECTED]
SIP/2.0Record-Route:
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To:
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 15:38:58
GMTCall-ID: [EMAIL PROTECTED]Supported:
timer,100relMin-SE: 1800Cisco-Guid:
2316671854-4109242839-3208043153-4243844325User-Agent:
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards:
9Remote-Party-ID:
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp:
1065195538Contact:
sip:[EMAIL PROTECTED]:5060Diversion:
sip:[EMAIL PROTECTED];reason=unconditionalExpires:
180Allow-Events: telephone-eventContent-Type:
application/sdpContent-Length: 332
v=0o=CiscoSystemsSIP-GW-UserAgent 8647 4690 IN IP4
213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio
16836 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101
0-16a=rtpmap:19 CN/8000
23 headers, 14 linesUsing latest request as basis requestSending
to 213.137.73.176 : 5060 (non-NAT)Found audio format 4Found audio
format 18Found audio format 101Found audio format 19Found
description format G723Found description format G729Found description
format telephone-eventFound description format CNCapabilities: us -
524302, them - 257/0, combined - 0Non-codec capabilities: us - 1, them -
3, combined - 1Sip read: INVITE sip:[EMAIL PROTECTED]
SIP/2.0Record-Route:
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To:
sip:[EMAIL PROTECTED]Date: Fri, 03 Oct 2003 15:38:58
GMTCall-ID: [EMAIL PROTECTED]Supported:
timer,100relMin-SE: 1800Cisco-Guid:
2316671854-4109242839-3208043153-4243844325User-Agent:
Cisco-SIPGateway/IOS-12.xAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFOCSeq: 101 INVITEMax-Forwards:
9Remote-Party-ID:
sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=offTimestamp:
1065195538Contact:
sip:[EMAIL PROTECTED]:5060Diversion:
sip:[EMAIL PROTECTED];reason=unconditionalExpires:
180Allow-Events: telephone-eventContent-Type:
application/sdpContent-Length: 332
v=0o=CiscoSystemsSIP-GW-UserAgent 8647 4690 IN IP4
213.137.65.234s=SIP Callc=IN IP4 213.137.65.234t=0 0m=audio
16836 RTP/AVP 4 18 101 19c=IN IP4 213.137.65.234a=rtpmap:4
G723/8000a=fmtp:4 annexa=noa=rtpmap:18 G729/8000a=fmtp:18
annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101
0-16a=rtpmap:19 CN/8000
23 headers, 14 linesIgnoring this requestLooking for 14103445557
in sipinboundRDNIS is 4103445557list_route: hop:
sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.176list_route:
hop: sip:[EMAIL PROTECTED]:5060Transmitting (no
NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To:
sip:[EMAIL PROTECTED];tag=as075e701dCall-ID: [EMAIL PROTECTED]CSeq:
101 INVITEUser-Agent: Asterisk PBXContact:
sip:[EMAIL PROTECTED]Content-Length: 0
to 213.137.73.176:5060 -- Executing
Dial("SIP/-0810da50", "Zap/5-1") in new stack -- Called
5-1 -- Zap/5-1 is ringingTransmitting (no
NAT):SIP/2.0 180 RingingVia: SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1Via:
SIP/2.0/UDP 213.137.65.234:5060From:
sip:[EMAIL PROTECTED];tag=16A2EDA0-5B2To:
sip:[EMAIL PROTECTED];tag=as075e701dCall-ID: [EMAIL PROTECTED]CSeq:
101 INVITEUser-Agent: Asterisk PBXContact:
sip:[EMAIL PROTECTED]Content-Length: 0
to 213.137.73.176:5060 -- Zap/5-1 is
ringing -- Zap/5-1 answered SIP/-0810da50We're at
[Asterisk-Users] asterisk and 3com
Hi! Anybody have experience using asterisk and 3com voip systems? Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error message 49159
Hi All I have that error message: WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) What can be the problem? Thanks! miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error message 49159
Hi Martin Please explain, why did you send the messages? miklos - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 2:04 PM Subject: Re: [Asterisk-Users] error message 49159 Martin Pycko wrote: We send SIP messages to that device up to 6-7 times and then we stop and this message shows on the console. WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) So it isn't really an error then, but an artifact of something asterisk is trying to do? I have seen these messages pretty much since the beginning of time, and I figured something was out of spec with my phones. I can't tell from what you say whether it is normal or not to see those messages? Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ERROR CODE
Hi! I have that message: *CLI WARNING[49159]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 177 (Request) I was thinking..why that call is for 127.0.0.1 is it the loopback of the asterisk machine? Thanks for any help Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I have a strange problem with ICH calls
Hi! I have a strange problem with ICH calls. When i try to make a call with asterisk for ICH nothing happens ( register is ok) But when i register my snom 200 with ich it works very well with the same register data. Someone knows anything about? miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I have a strange problem with ICH calls
Ok
extensions.conf:
exten = _7X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
sip.conf:
register =31451543:[EMAIL PROTECTED]/33
[iconnect]
type=friend
secret=
username=31451543
host=sipauth.deltathree.com
dtmfmode=inband
context=from-sip
miklos
- Original Message -
From: Andrew Joakimsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 30, 2003 5:27 PM
Subject: RE: [Asterisk-Users] I have a strange problem with ICH calls
Please post your extensions.conf and sip.conf sections relevant to
ich/deltathree.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of listas iPfone
Sent: Tuesday, September 30, 2003 3:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] I have a strange problem with ICH calls
Hi!
I have a strange problem with ICH calls.
When i try to make a call with asterisk for ICH nothing happens (
register
is ok)
But when i register my snom 200 with ich it works very well with the
same
register data.
Someone knows anything about?
miklos
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[Asterisk-Users] ICH PROBLEM
Hi !
I´m using * with a snom 200 phone, i can use FWD but cant use ICH.
Someone can tell me if my setup is correct?
sip.conf:
register =user:[EMAIL PROTECTED]/33
extensions.conf:
exten = _7X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
In CLI the registration is ok but when i try ex. 7551136752312 nothing
happens, i get a forbiden message.
Thanks
Miklos
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 29, 2003 8:26 AM
Subject: Re: [Asterisk-Users] IAX and NAT
Brancaleoni Matteo wrote:
VoIP protocols normally use 2 connection:
* 1 for control (eg on port 5060 for sip)
* 1 for the RTP (media stream)
The latter hasn't a fixed port, since is negotiated
by the control connection. That could cause some troubles
with NAT firewalls.
IAX doesn't use 2 ports, but only one .
So on the same port it brings the control connection
the RTP stream. So NATting IAX isn't a problem
Also, IAX is client-driven, the IAX client opens a channel to the server
and
keeps it open for calls both ways.
SIP is a peer-to-peer protocol and a phone needs to be able to receive
incoming
calls. If the phone, or the SIP UAC/S (User agent client/server)
software,
is behind a NAT, there's no way any phone out there can reach it on the
inside.
There are a lot of fixes, ranging from using a SIP proxy on the outside
for incoming
calls and keeping a NAT session open with fixes called NAT pings to
protocols
that opens up port forwarding from the NAT to the inside client (UPNP) and
protocols
that let the client investigate the NAT situation (STUN) and be more
clever.
The long term fix is to remove NAT boxes and use IPv6 or allocate more
IPv4 addresses
...or, as some people on this list advocate, use another protocol.
/Olle
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Re: [Asterisk-Users] CDR Web Search Frontend
Oi Adriane! Minha mãe foi internada hoje de madrugada no 9 de julho por causa de um problema de estomago.. Já viu que não vou conseguir ir hoje tb. Já estou de pé desde ontem a noite. arrumei um micro aqui no hospital para te escrever. esqueci meu celular em casa. Amanhã ainda dá tempo né? eu não vou deixar a sua chefinha quebrar o seu pescocinho tá? um beijo claudio - Original Message - From: PJ Welsh [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, September 29, 2003 9:52 AM Subject: Re: [Asterisk-Users] CDR Web Search Frontend On Mon, Sep 29, 2003 at 11:09:06AM +0100, WipeOut wrote: I was thinking of using http://developer.berlios.de/ As SF has had many problems recently :( Regards Mark Yea, I have noticed Sourceforge has been a little flaky lately.. Thought they would have been on top of it quicker.. http://developer.berlios.de/ seems to be down for me in the US at this time: * Connection Failed The remote host or network may be down. Please try the request again. Generated Mon, 29 Sep 2003 12:51:36 GMT... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ERROR MESSAGE
Hi! Thaanks the problem was the same, now i´m using a static ip and all is working fine. regards - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 4:07 PM Subject: RE: [Asterisk-Users] ERROR MESSAGE I had this problem when I changed the IP of one of the * boxes. Did not see it on the other boxes. Have you changed the IP of your * box since compiling * first time? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie: Crossing my fingers
That really help me: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+files miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
I´m doing the same, ix66 asterisk. Did you registered asterisk in the ix66? Please share your set up, i´m with some truble using ICH . Miklos - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 2:03 PM Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration On Thu, 2003-09-25 at 18:54, Michael Koehler wrote: A plain wireless dlink dsl router. I'm testing one of these http://www.intertex.se and my * is behind it. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ERROR MESSAGE
Hi I have that error messages, what does it mean? *CLI WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Request) miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] RE: [Asterisk-Users] can´t call ICH
Hi!
There is my sip.conf:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
maxexpirey=180 ; Max length of incoming registration we allow
defaultexpirey=160 ; Default length of incoming/outoing registration
disallow=all
allow=gsm
allow=ulaw
allow=alaw
tos=reliability
register =user:[EMAIL PROTECTED]/33
register =user:[EMAIL PROTECTED]/33
register =user:[EMAIL PROTECTED]/33
[fwd]
type=friend
secret=ipfone001
username=400277
host=fwd.pulver.com
context=from-sip
[welcome]
type=friend
secret=welcome
username=5
host=fwd.pulver.com
context=from-sip
[iconnect]
type=friend
secret=3587
username=31451543
host=sipauth.deltathree.com
dtmfmode=inband
context=from-sip
[33]
type=friend
secret=33
username=33
host=dynamic
defaultip=192.168.0.31
dtmfmode=rfc2833
mailbox=331
context=from-sip
callerid=snom200 33
[34]
type=friend
secret=34
username=34
host=dynamic
defaultip=192.168.0.36
dtmfmode=rfc2833
mailbox=331
context=from-sip
callerid=snom100 34
[35]
type=friend
secret=35
username=35
host=dynamic
defaultip=192.168.0.33
dtmfmode=rfc2833
mailbox=331
context=from-sip
callerid=ipdialog 35
- Original Message -
From: Paul Crick [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 23, 2003 7:19 PM
Subject: [Asterisk-Users] RE: [Asterisk-Users] can´t call ICH
You've got a whole bunch of numbers you're trying to call there. What is
the
full number that you want to call including the country code? It's not
clear
if the number you're trying should be 755xxx 55xxx or 055xx ?
and my extensions:
[from-sip]
exten =33,1,DIAL(SIP/33,20,tr)
exten =34,1,DIAL(SIP/34,20,tr)
exten =35,1,DIAL(SIP/35,20,tr)
exten = _9x.,1,DIAL,Zap/g1/${EXTEN:1}
exten = _8X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _7X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten =331,1,VoicemailMain,s331
The number i want to call is 55 11 36752312
I hope this helps.
Thanks !
Miklos
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[Asterisk-Users] error message playing .mp3
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Re: [Asterisk-Users] error message playing .mp3
Thanks Gavin! It works now. Miklos - Original Message - From: Adams, Gavin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 9:51 AM Subject: RE: [Asterisk-Users] error message playing .mp3 -Original Message- From: listas iPfone [mailto:[EMAIL PROTECTED] Somebody knows why asterisk gives me that error wile playing .mp3 files? The files play well but the message aperas any way: WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0 of 4 bytes) (No such file or directory)! Listas, You might try down-sampling the MP3 files to 160Kb/sec, mono through LAME or some other MP3 encoder. Prior to converting some royalty-free music from 320Kbs joint-stereo, mpg123/asterisk would barf on the file. Assumeably due to the original encoding (EAC under Windows/LAME). HTH, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Fw: hangup problem Brazil
- Original Message - From: iPfone Telefonia IP To: [EMAIL PROTECTED] Sent: Friday, September 19, 2003 11:27 AM Subject: hangup problem Brazil Hi all!I´m setting up an asterisk box here in brazil, asterisk don´t hangup afterthe caller disconects...it goes to voice mail etc.. Somebody have the sameproblem?I received that advice from digium support but it dont works:Edit the file "dsp.c" which is in your asterisk source. At the top ofthe file find "#define DEFAULT_THRESHOLD 1024" and change the 1024 to128. Find the "#define BUSY_MIN 75" and change the 75 to 65. Find"#define BUSY_MAX 1100" and change the 1100 to 200. Save the file. Thendelete the file "dsp.o" and then do a "make install". Then reload themodules and start asterisk.When i put callprogress=yes in the conf file the sistem don´t answer thecalls any more, like another postings here.Busydetect=yes dont makes any diference, dont works to...Regards for allMiklos
