[asterisk-users] SIP Trunk - problem to connect

Hello! Thnxs for reading! I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset, for instance (and it works!) Connection parameters are: Authentication Name: Número 11 Authentication password: 12345678 Username: 11 Display name: 11 Domain: hpbx.iplanne

[asterisk-users] Can not calculate far_max_ifp before far_max_datagram has been set

Marco-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To

[asterisk-users] Asterisk 13, PJSIP and T38 problem

t38_udptl=yes t38_udptl_ec=fec t38_udptl_maxdatagram=400 [trunk-patton] type=auth auth_type=userpass password=X username=X = Thanks Marco -- _ -- Bandwidth and Colocation Pr

[asterisk-users] Debugging issues with setup

Hello, I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a previous server. I have replicated all the configurations, modules and setup that I know of. However, when I tested an outbound call, it didn’t work. Checking the asterisk message log yielded nothing. Any idea

[asterisk-users] R: Asterisk and Call Hold

risk 11, but there is the same problem. I've already read all the information about canreinvite and directmedia Can anybody help me? Thanks a lot Marco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] WSS over Asterisk

where the SIPML5 seems not able to connect to the asterisk box. Thank you and best regards, Marco Signorini. On 06/12/2014 03:21 AM, Steve Ng wrote: I am using Asterisk v12.3. As far as DTLS, I understand that applying the following Javascript will temporarily fix for SIPML5 to Asteri

[asterisk-users] sipML5, Ast12 and WebRTC: not acceptable here

the right direction? Below is my configuration. The sofpthone is registered as 1060. Thanks in advance. Marco Signorini. pjsip.conf: [transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/etc/asterisk/sslcert.pem method=tlsv1 [1060] type=endpoint transport=transport-tls context=from-i

Re: [asterisk-users] DAHDI and Oslec

#x27;s very old so I can't tell you if this is something true for Debian 6.06 too. Thanks. Marco Signorini. On 02/26/2013 05:38 PM, Doug Lytle wrote: I'm hoping someone can help me here. I've purchased replacement systems for 3 aging 1.4.x installs. I'm hoping to setup

[asterisk-users] Call Hold problem

lds is present the local ip address and not the next hop ip. This is the log : http://pastebin.com/ARUC0j4t The asterisk IP : 87.248.56.101 The next hop IP : 87.248.56.100 Is it a bug? i'm already search on google, but i dont find anything. Let me

[asterisk-users] R: R: R: Asterisk and History-Info

Ok, thanks for all Best Regards Marco -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Joshua Colp Inviato: mercoledì 26 settembre 2012 19:37 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto

[asterisk-users] R: R: Asterisk and History-Info

IONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "+39zzz" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 239 Thanks a lot! Marco Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-

[asterisk-users] R: Asterisk and History-Info

oun...@lists.digium.com> [mailto:asterisk-users-boun...@lists.digium.com]<mailto:[mailto:asterisk-users-boun...@lists.digium.com]> On Behalf Of Marco Colombo Sent: Wednesday, September 26, 2012 10:33 AM To: Asterisk-Users Subject: [asterisk-users] Asterisk and History-Info Hi All,

[asterisk-users] Asterisk and History-Info

Hi All, Someone can tell me if asterisk support the SIP History-Info? If it supports, how can enable it? I searched on Google, but I could not find anything... Thanks for all Best Regards MC

[asterisk-users] R: SIP CANCEL, Reason

[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matthew Jordan Inviato: giovedì 20 settembre 2012 13:42 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] SIP CANCEL, Reason - Original Message - > From: "Marco Colombo" &

[asterisk-users] SIP CANCEL, Reason

Hi All! i have a problem with asterisk 1.8.11. I must have in the SIP cancel message, the line "Reason" Example : Reason : SIP;cause=16;text="Normal Call Clearing" I have already enable "use_q850_reason=yes", but this not work. In my dialplan I have already add : exten => _X.,n,Hangup(${HANGUPCAU

[asterisk-users] Asterisk 1.8 - BRI D Channel going up and down every few seconds

, version of LibPRI etc. has anybody experienced these problems on BRI? Any suggestions with regards to these warnings are welcome! Kind regards, Marco Mooijekind. -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Problem installing B410P BRI card for asterisk

dahdi_tool[40+3000] And a lot of "Wrote 0x0 to register 0x1ab but got back 0x4" statements. If i run dahdi_tools it fails with a segmentation fault. Any suggestions are appreciated! Kind regards, Marco Mooijekind. -- ___

Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Hello Gord, the line icon is solid black, which should indicate the lines are registered. Marco. On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart wrote: > Does the phone show the line as registered? The little phone icon on the > display should be solid for a registered line and

[asterisk-users] Dialing problem with Polycom phones after SIP update

s problem , any tips! Thanks in advance! Marco Mooijekind. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.ast

Re: [asterisk-users] How to program a 100ms delay between the ringing of queued calls w/ ringall

Maybe local channels will do the trick? They allow you to schedule delays between subsequent devices ringing. Not sure whether they work as queue members.. Marco. Op 5 dec. 2011 16:35 schreef "Sammy Govind" het volgende: > Hi, > I dont think that 2 Queue commands would help, als

Re: [asterisk-users] Issue with Polycom SPIP650 and its sidecar

Maybe use a power supply instead of PoE, see if problem still occurs. Marco. Op 30 nov. 2011 18:46 schreef "Olivier" het volgende: > > > 2011/11/30 Mike > >> Hi Olivier, >> >> ** ** >> >> It if occurs only on the sidecar, I would imagine

Re: [asterisk-users] USB or Ethernet based FXO device ?

? We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in Italy. My concern is about reliability of USB Any success stories with it? Tips and tricks? Thank you and regards, Marco Signorini. -- INGEGNI Tech S.r.l. site http://www.ingegnitech.com mail i...@ingegnitec

Re: [asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom

address associated to the Ethernet shield on top of Arduino. Thanks, Marco Signorini -- http://www.ethermania.com http://www.ingegnitech.com David - asterisk list wrote: > Asterisk as a phone system makes perfect sense in a condo. You can get > > all the DID's you want and elimina

[asterisk-users] Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone

or helping me gettng started with asterisk Marco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk

Re: [asterisk-users] Asterisk to switch on electric heaters remotely?

Hi Did you looked at Arduino + Ethernet Shield? Is something you can program in C or C++ to receive a simple TCP and/or HTTP packet and turn on an external relay. >From the dialplan you can run an http query through curl and/or an external AGI command. Best regards, Marco Signorini. -- Ma

[asterisk-users] Sangoma A500 NT BRI PTMP without woomera on asterisk 1.6

Hello I recently heard this should be possible. Has anyone experience with this? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every T

Re: [asterisk-users] Problems with Dahdi 2.3.0.1 trying to load OSLEC

Best regards, Marco Signorini. -- = - http://www.ethermania.com - - http://www.ingegnitech.com - Jose P. Espinal wrote: > Hello list, > > > I'm facing a little issue with dahdi attempting to load the OSLEC echo > canceller into my current ke

Re: [asterisk-users] SuSE Firewall2 - Port Forward Command

SEfirewall2 For example: FW_FORWARD_MASQ="0/0,192.168.10.1,udp,5060,80,192.168.2.3" lets you able to forward the udp 5060 from the IP 192.168.10.1 to 192.168.2.3 You need to add all the other RTP relevant rules. Best regards. Marco Signorini -- = EtherMa

Re: [asterisk-users] audio glitches in conference

n either look at title of its span in /proc/dahdi file for a "source:" in the description. Or even run: strings dahdi.ko | grep source: -- Marco Mouta On Thu, Feb 25, 2010 at 8:15 AM, wrote: > It looks to me that u are having clock synchronism problems due to the fact > you are

Re: [asterisk-users] audio glitches in conference

't new and is well known due to the fact u don't have a precise clock source for meetme.. You need to have chan_dahdi dummie... Hope it helps. Marco Mouta Enviada do dispositivo sem fios BlackBerry® -Original Message- From: "Jeff Brower" Date: Wed, 24 Feb 2010 18:2

Re: [asterisk-users] verifying correct loading of VPMADT032

etter configuration was found. Regards, Marco. -- http://www.ingegnitech.com http://www.ethermania.com Greg Woods wrote: > On Sat, 2010-01-02 at 20:25 +0100, F6HQZ wrote: > > >> cat /proc/interrupts >> Search the Digium cards drivers and look if several interfaces are usi

[asterisk-users] 1800 DID Provider - Suggestion

Hello All, Do you guys suggest any 1800 DID Provider in the US ? I'm having a hard time to find one. Thanks, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or u

[asterisk-users] lawnmower man "attack" ??

beginning…. Looking forward to hearing from you guys ;) Cheers, -- Marco Mouta ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

[asterisk-users] lawnmower man "attack" sip tag=Zerogij34 some one else notice this in 20th september or recently?

beginning…. Looking forward to hearing from you guys ;) Cheers, -- Marco Mouta ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.as

[asterisk-users] Asterisk Queue & Agent

with agent id and a predefined postfix file name *SECOND QUESTION*: how can I set the queue to play an estimated hold time in queue to the member in the queue I can play only to agent. Someone can help me Thanks to all for your help Marco __

[asterisk-users] SIP doesn't recognize hangup

ine But if the external caller hang up the call ... the call to NUMBERTOCALL on acc1 continue to ring until the called answer, but the call is out. Someone can help me ?!?!? Thanks to all Marco ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Asterisk 1.4.25 and attended transfer

Just done it ... and all works fine. Thanks all. Marco 2009/7/24 Administrator TOOTAI > Marco Sambo a écrit : > > Hi all, > > I've a problem: I update my asterisk to version 1.4.25, and the attended > > transfer doesn't work. > > > [...] > &

[asterisk-users] Asterisk 1.4.25 and attended transfer

all return to A. CORRECT if B hangup, .. also the call hangup Someone can help me??? Please! Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update opt

Re: [asterisk-users] how to use patgen and pattest for PRI card?

l and to http://lists.digium.com/pipermail/asterisk-dev/2009-March/037003.html Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Chris YM wrote: > hello: > I wan to use the test tools-patgen and pattest for pri cards. > according to

[asterisk-users] USB phone with Asterisk under Linux

Hi all, I want to try to use a USB phone with Ekiga under Linux (Debian Lenny). It works: I can receive and make calls. But some buttons of USB phone don't work properly. In particular, button *, #, and hangup have wrong key mapping. Someone have tried a USB phone Thamks all

Re: [asterisk-users] Echo and static on PRI with errors

than > .. > bridge architecture might be forcing interrupts from some cards > to use a single line/IRQ. > > > Thank you for your complete description on how PCI IRQ subsystem works. It's probably the best explanation I've found since years. My warm comp

Re: [asterisk-users] Echo and static on PRI with errors

ts impossible to change the IRQ assignment for expansion cards. This is not always true and sometimes swapping add-on cards solves the problem. We had better results with cards based on new Digium technology or with Sangoma cards. Best regards, Marco Signorini. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Click-to-dial CTI for Windows

Hi, I try Noojee Click and Outcall, and for my context they work fine. Some times ago I tried SanpANumber, but it was bought by Digium and substitute with ADA. Bye Marco 2009/6/15 Stefanov, Milen > Hello guys, > > Is there a decent click-to-dial CTI which works well with Aster

[asterisk-users] RES: RES: SIP Response 181 - Is it possible in A steri sk?

to it. But does any one have a suggestion, or real scenario similar to this that could help me?? Thanks again, Marco -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen Enviada em: terça-feira, 2 de

Re: [asterisk-users] Play a file while transfering a call

Hi, I do this by creating a directory "waitingtransfer" with only 1 file (the audio message, the name isn't important, so you can change it everytime you want) and then add new musiconhold class with specific "waitingtransfer" directory. In your extensions.conf you change the musiconhold class to w

[asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?

Hi Philipp, So, what you are saying is that SIP trunks between 2 Asteriks might be able to handle SIP Response 181 ? Marco -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen Enviada em: terça-feira

[asterisk-users] RES: SIP Response 181 - Is it possible in Asteri sk?

Thanks Philipp, Sorry about my ignorance, but what would be IIRC Asterisk Trunk? Where could I find info about it? Thanks again, Marco -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen Enviada

[asterisk-users] SIP Response 181 - Is it possible in Asterisk?

ng set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. Thanks, Marco Cordeiro mhcorde...@gmail.com ___ -- Bandwidth and Colocation Provided b

Re: [asterisk-users] CAll-limit or incominglimit ?????

Hi, in Asterisk 1.4 to limit the simoultaneous calls I use the following parameters: [general] ... limitonpeers=yes notifyringing=yes [phone] ... host=dynamic username=phone call-limit=2 So I can receive and make max 2 calls simoultaneous. Fo me that's work fine. 2009/5/29 Yuri > Good m

Re: [asterisk-users] SIP over VPN

:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo > *Sent:* Tuesday, May 26, 2009 11:21 AM > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] SIP over VPN > > > > Hi all, > I have a questio

[asterisk-users] SIP over VPN

Hi all, I have a question. I have a VPN and I want to use a SIP softphone on my notebook using with asterisk. But I have some problem with firewall and port. Someone knows which ports I should open on my firewall??? I can't connect ... Thanks all.

Re: [asterisk-users] h extension and channel variables

I set a variable CalledID to ${EXTEN} before dial it. So in h extension I can use ${CalledID}. 2009/5/26 Thomas Kenyon > On 5/26/2009 10:57, Thomas Kenyon wrote: > > Is there a method to fetch the ${EXTEN} of the channel that has been > > hung up when exten h is started? > > > > The nearest

Re: [asterisk-users] Problem with Asterisk + TDM410 FXO

FXO channels shuld have FXS signalling, and FXS channels shuld have FXO signalling, so: # FXO channels are 1,2,3 fxsks=1,2,3 # FXS channel is 4 fxoks=4 > sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is > > that a attached fxs presents internally as a fxo > > > > I have

[asterisk-users] High Volume US Traffic? Claim DIP Compensation!

gards, Marco Wind dipfees.com Ph: 646-736-7816 Tf: 888-780-0253 F : (347) 626-2242 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] FOP and UserEvent()

On client I connect to FOP panel, but I don't see any popup. Someone can help me to configure FOP popups and in the use of UserEvent() application? Thanks all Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Asterisk and HUD server

Well, I see the rpm but my Asterisk box has Debian Linux, and I'm a little afraid to use alien package to transform rpm to deb. Has HUDlite Server source?? Like in tar.gz?? 2009/4/23 David Klaverstyn > Hi Marco, > > > > Try this: > http://yum.trixbox.org/centos/4/RPMS/

[asterisk-users] Asterisk and HUD server

Hi, someone has installed on an Asterisk box (not Trixbox) with Debian Linux, the HUDlite Server? Can someone help me in retrieve and install packages??? Thanks all Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3

Lee Howard wrote: > Marco wrote: > >> I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine. >> They are linked together through localhost. I've turned qualify on for the >> iax peer. Periodically I've this message: >> >&g

Re: [asterisk-users] Asterisk process ended

Hi, I have the same problem: sometimes my Asterisk box crash (or similar) and in asterisk log doesn't appear nothing. Also into syslog. I don't understand what is it!!!! Marco 2009/4/21 Adrien Lemoine > Hi all, > > > > I experienced for a second time the crash o

[asterisk-users] Peer 'iaxfax' is now UNREACHABLE! Time: 3

I can do to better understand what's the cause of this? Thank you and best regards, Marco Signorini. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Remote BLF / hint on IAX2 trunk

So thanks, but in Asterisk 1.4.24 is not present in any way?? Any mystique solution?? Marco 2009/4/16 Tilghman Lesher > On Thursday 16 April 2009 07:08:49 Marco Sambo wrote: > > Well, I use SIP phones and IAX2 trunk. For the moment I don't want to use > > SIP trunks.

Re: [asterisk-users] Remote BLF / hint on IAX2 trunk

Well, I use SIP phones and IAX2 trunk. For the moment I don't want to use SIP trunks. Can you help me? 2009/4/16 Philipp Kempgen > Marco Sambo schrieb: > > I have a question: how can I see hints of a remote Asterisk in IAX2 > trunk?? > > I want to set BLF on my phones

[asterisk-users] Remote BLF / hint on IAX2 trunk

Hi all, I have a question: how can I see hints of a remote Asterisk in IAX2 trunk?? I want to set BLF on my phones to look state of other phones also in other Asterisk server. Someone have any idea or solution? I use Asterisk 1.4.24. Thanks all Marco

Re: [asterisk-users] TDM2400P dial tone is not present on phones, but the phone ring with incoming calls

Hi, excuse me, but I see in your code that you configure DAHDI with OSLEC. How do you do? Which version you have installed? Thank you. Marco 2009/4/16 Giovanni Magallanes > Hi, > > I have a problem with TDM2400P card. The card is detected ok, I can make a > call but onl

[asterisk-users] Asterisk and Voice Recognition Sphinx

Hi all, someone has used the voice recognition software named Sphinx??? Can he tell me how to use and its performance??? Thanks Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Logging Asterisk console

tagidir => /var/lib/asterisk/agi-bin astspooldir => /var/spool/asterisk astrundir => /var/run/asterisk astlogdir => /var/log/asterisk [options] verbose = 3 and so I find into /var/log/asterisk the logpro file with the output of CLI (verbose) and notice, warning, error, debug messag

Re: [asterisk-users] DAHDI with OSLEC

One thing! I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel 2.6.28 or newer to use oslec with DAHDI??? 2009/4/1 Marco Sambo > But I don't have also echo > > modinfo echo > modinfo: could not find module echo > > > > > > 2009/

Re: [asterisk-users] DAHDI with OSLEC

But I don't have also echo modinfo echo modinfo: could not find module echo 2009/4/1 Dave Fullerton > Marco Sambo wrote: > > Mhmm. Thaht's strange! > > > > modinfo oslec > > --> > > modinfo: could not find module oslec &

Re: [asterisk-users] DAHDI with OSLEC

per depends:dahdi vermagic: 2.6.26-1-486 mod_unload modversions 486 2009/3/31 Tzafrir Cohen > On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote: > > Hi, > > I've a problem: I can't configure DAHDI with ech canceller OSLEC. > > I have Asterisk

[asterisk-users] DAHDI with OSLEC

an help me? Thanks Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Ebay's SIP for Skype

I have to try Skip2PBX, integrated into my Asterisk machine, but it seem more invasive than Gizmo5 opensky. Doesn't it? Marco 2009/3/26 Grygoriy Dobrovolskyy > skip2pbx is the best i tryed, but nasty price ;) > > ___ > -- Bandwid

Re: [asterisk-users] Ebay's SIP for Skype

Well, anyone knows a good Skype vs SIP channel or program or something else to integrate it into an Asterisk machine, to call normal skype users and not and receive normal skype calls? I red that Digium and Skype are working to integrate a chan_skype. Anyone can tell me about? Bye Marco 2009/3

Re: [asterisk-users] 428 Loop Detected

It's so uncommon for me fxs and fxo cards and based on the reference of sip.conf files and accounts i totally missed last paragraph where it was mentioned only analogue lines and fxs (phone). my appologies. E1 and BRIs and sip trunks have been overloading my last month of work. cheers, --

Re: [asterisk-users] Busy on SIP

ame=10 secret=1234 amaflags=documentation accountcode=sip10 callerid="sip10" <10> call-limit=2 dial=SIP/10 canreinvite=no And this resolve for me problems for busy and for xfer Aastra button. Marco 2009/3/17 Ira > At 01:29 AM 3/17/2009, you wrote: > >But there is anothe

Re: [asterisk-users] Busy on SIP

Ok, I read it. Thank u. For busy on SIP I use also the Asterisk peer function SIPPEER with field CURCALLS. 2009/3/17 Philipp Kempgen > Marco Sambo schrieb: > > Anyone know how to use busy-level parameter or some other useful > parameters? > > call-limit=2 > busy-level

Re: [asterisk-users] Busy on SIP

eful parameters? Thanks all Marco 2009/3/16 Gordon Henderson > > On Mon, 16 Mar 2009, Olivier wrote: > > > 2009/3/16 Gordon Henderson > > < > gordon%2baster...@drogon.net > > >> > > > >> On Mon, 16 Mar 2009, Marco Sambo wrote: > >> &g

[asterisk-users] Busy on SIP

x27;t understand why I find it avaible when it makes an outgoing call. Thanks all Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digiu

Re: [asterisk-users] 428 Loop Detected

n => s,1,Dial(SIP/phone,10) exten => s,2,Voicemail(line) exten => s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal wrote: > Hi I looked at few email

Re: [asterisk-users] Faxing success rate on PRI

Thank you, Doug, for precious information. Best regards, Marco Signorini. === INGEGNI Tech S.r.l. http://www.ingegnitech.com Doug Lytle wrote: > > Main fax server: > > > Mandriva 2008.1 > Kernel 2.6.24.5 (Compiled for source) > (1) Intel(R) Xeon(TM)

Re: [asterisk-users] Faxing success rate on PRI

ax servers. Thank you for writing SpanDSP and best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Steve Underwood wrote: > Marco Signorini wrote: > >> Thank you to All People answered me on this subject. >> Analyzing your answ

Re: [asterisk-users] Faxing success rate on PRI

Thanks Doug and Lee, your testimonials are changing my opinion :-) Can you provide some details about your setup? What PRI solution are you using? And what version of Asterisk, IAXModem, SpanDSP? Thank you and best regards, Marco Signorini === INGEGNI Tech S.r.l. http

Re: [asterisk-users] Faxing success rate on PRI

rt. Thank you to All People answered me on this subject. Analyzing your answers, seems that fax handling is still today problematic with IAXModem and Hylafax... or I'm wrong? What about other solutions? Thank you and best regards, Marco Signorini === INGEGNI T

[asterisk-users] Faxing success rate on PRI

in this list in the past, but I would like to know if someone has experience on this and could share their opinion, tricks and/or statistical results on failure/success rate when faxing. I think that this could be useful to other people have to realize a system like that one depicted. Thank you in

Re: [asterisk-users] faxing via linksys SPA3102 half page goes through

Joseph wrote: > On 03/04/09 15:44, Marco Signorini wrote: > >> Hi Joseph. >> I've spent some time tuning the SPA3102 FXS line input and output gain >> and I think that this is an important variable. >> Let's try to record incoming and outgoing f

Re: [asterisk-users] faxing via linksys SPA3102 half page goes through

e recorded file with a wave editor (Audacity). I had better results if the maximum level is near half to the full dynamic. Then switch to T38, if you need it. Hope this helps you. Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Joseph wrote:

[asterisk-users] patlooptest and TE121P

found on http://kb.digium.com/entry/138/ for the E1. Thank you and best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-use

Re: [asterisk-users] building a phone

Jon Pounder wrote: > Marco Signorini wrote: > >> It's a dream! >> It's since years that I'm thinking to have an open hardware project >> targeted to a SIP application. >> >> > > there is already a project called openmoko - join it

Re: [asterisk-users] building a phone

ilding/desktop music streamer, or SIP compliant actuators. I have a (very) little experience on electronic projects. Is there something I can do to help starting a similar project? Thank you and best regards. Marco Signorini Tzafrir Cohen wrote: > Hi folks > > A common wisdom here is

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

quot;. Best regards, Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com Tiago Durante wrote: > On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos wrote: > >> Marco Signorini wrote: >> >>> Hi Tiago. >>> >>>

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

Hi Tiago. I've it working on PHP 5.2.6 but only after having modified the php.ini default configuration keys: zend.ze1_compatibility_mode = Off short_open_tag = Off setting together to On and restarting apache forces PHP5 to behave like PHP 4.x version. regards, Marco Sign

Re: [asterisk-users] Siemens Hipath PRI to Asterisk Call Routing?

try to set in your zapata.conf overlapdial=yes then in your asterisk cli reload chan_zap.so -- Marco Mouta On Fri, Feb 13, 2009 at 9:21 AM, wrote: > Default FreePBX context, > > [from-pstn] > include => from-pstn-custom ; create

[asterisk-users] Suggestion for a new server for E1 line

r for one humanitarian organization in Italy. Any suggestion is really welcomed. Thank you very much. Best regards, Marco Signorini http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-dig

Re: [asterisk-users] Dahdi Init script for Suse?

rks for me, except, if I remember well, some little problem on reload (but stopping and starting again works fine). Best regards, Marco Signorini. == INGEGNI Tech S.r.l. http://www.ingegnitech.com > Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10

Re: [asterisk-users] Portech MV-378 with Asterisk

Yes. That's the correct way to do it. Placing # as a rule in callnum forces the Portech to use the number defined in the SIP INVITE packet. Bye. Marco. Marco Signorini INGEGNI Tech S.r.l. http://www.ingegnitech.com <http://www.ingegnitechcom/> Pascal

Re: [asterisk-users] Portech MV-378 with Asterisk

Pascal Bruno wrote: > Thanks for your reply! > > Can you tell me what you have in your Portech configuration settings > (Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file > is pretty similar to yours but still cant register. > > > > On Fri, Jan

Re: [asterisk-users] Portech MV-378 with Asterisk

I remember something wrong in cryptography chiper/dechiper based on realm... So, if you have problems, let's try to specify the asterisk raw IP address in the Portech. Best regards, Marco Signorini. ___ -- Bandwidth and Colocation Provided by http://w

Re: [asterisk-users] Oslec issue

e not working on systems where more than one zap channel was present and I was not able to test it on these type of situations. Thank you and bye Marco Signorini Joseph L. Casale wrote: >> I spent some time to understand what's missing in the OSLEC patch for >> dahdi... I can

Re: [asterisk-users] Oslec issue

trunk. I never did a commit on asterisk svn so I need some hints on how to do it. Thank you and best regards. Marco Signorini. Joseph L. Casale wrote: > Yesterday I pulled in the latest svn of Dahdi and added the files > from a recent kernel in the drivers/staging/echo structure and modified

[asterisk-users] Persistentmembers (Not working with restart)

Thanks, Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problem with DAHDI and OSLEC integration.

the svn revision 5366 into my temporary folder /home/marco/Install/dahdi-linux 2. Taken the linux-2.6.27 kernel sources baseline and placed in my temporary folder /home/marco/install/linux-2.6.27 3. Taken the Linux kernel patch-2.6.28-rc6.gz, unzipped and applied to the baseline kernel 2.6.27. This

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