Re: [asterisk-users] Asterisk Outlook integration

Hi BB, you could try this: http://asterisk-outlook-dialer.voip-singapore.qarchive.org/ Never tested it deeply but apparently seems to work fine. Giorgio Incantalupo Bruce B wrote: Hi Guys, What is out there other than OutCall that works with M$ Outlook and Asterisk 1.4/1.6 ? I prefer

[asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)

-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo

Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)

: No translator path from alaw to unknown which are quite annoying...aren't they? Should I pay to avoid a CLI message? That doesn't sound fair to me. I know I should report the problem but the fake codec seemed the faster way. Giorgio Incantalupo Giorgio Incantalupo wrote: pbx18*CLI module load

Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)

for a different version module instead of asking for a license. Should I report this weirdness? Btw thank you for your time. Giorgio Incantalupo P.S.: as I've already written in some other post, I use the criminal codec to test Voip lines without the need to install the license every time

[asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk

Hi all, I've got 4 actors on my stage: Alice calling from outside Bob transferring incoming calls to Charlie Charlie who has a mobile phone My PBX which is connected to my ISDN line. I want Charlie to see Alice's Callerid after Bob has transferred the call as if Charlie is receiving the call

Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk

Hi Gopalakrishnan A.N, I tried it but it seems like my telco is overwriting the value I set as callerid. Maybe it is possible with Voip providers only. Giorgio Incantalupo Gopalakrishnan A.N wrote: Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I disabled the caller-id

Re: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Friday, November 19, 2010 9:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk Hi all, I've got

Re: [asterisk-users] Error with Connecting Two Asterisk BOX with IAX

Hi bakko, just as a test, try to add calltokenoptional=0.0.0.0/0.0.0.0 to your iax.conf. Giorgio Incantalupo bakko wrote: Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register = coiax:pa...@69.164.207.166 [smiax

Re: [asterisk-users] ztdummy IVR no voice

Hi Jordan, it happened to me too. I disabled the ACPI function and it worked. Giorgio Incantalupo jordan pan wrote: Hi all , In my pbx ,there is no zaptel card ,so i loading ztdummy,but problem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp

Re: [asterisk-users] asterisk and cisco 2800

it on the Cisco, type clock source internal under the controller config. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Friday, July 09, 2010 4:24 AM To: Asterisk Users Mailing List

Re: [asterisk-users] asterisk and cisco 2800

) there is nothing on internet about connecting asterisk and cisco... :( Giorgio Incantalupo Peder wrote: That's not right. Should be 1245 - 4512: http://www.voip-info.org/wiki/view/crossover+T1+cable -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk

Re: [asterisk-users] asterisk and cisco 2800

cable too: 1 -- 5 2 -- 4 4 -- 2 5 -- 1 and it seems to work because I get all alarms off after plugging it in. Thank you Giorgio Incantalupo Neeraj Chand wrote: Hi Giorgio, Why don't you terminate calls on the cisco router via SIP? -- Message: 11 Date: Fri

Re: [asterisk-users] asterisk and cisco 2800

Hi Peder, I'make a new cable following the info on that webpage. I hope it works with Cisco 2800 too! :) Thank you! Giorgio Incantalupo Peder wrote: That's not right. Should be 1245 - 4512: http://www.voip-info.org/wiki/view/crossover+T1+cable -Original Message- From

[asterisk-users] asterisk and cisco 2800

-1024' [Jul 2 15:20:36] VERBOSE[15031] logger.c: -- Hungup 'IAX2/1-1024' Any hints? Thank you. Giorgio Incantalupo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] sip_xmit: sip_xmit returned -1: Operation not permitted

Hi Jonas, I get this error when I incorrectly set my PBX gateway AND I have a sip peer trying to register outside (i.e.: a sip provider). Are you sure about your sip.conf? Giorgio Incantalupo Jonas Kellens wrote: Hello, my Asterisk CLI is flooded with the following message : [Jun 25 21

Re: [asterisk-users] Very strange registration problem

Jonas, have you checked the log files? Giorgio Jonas Kellens wrote: Hello list, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's

[asterisk-users] shared lines (sla) with Asterisk 1.4.26, any hints?

Hello, I'm trying to setup shared lines with Asterisk 1.4.26 and Snom phones. It seems that Asterisk works correctly (I get State: SLA_TRUNK_STATE_RINGING from the CLI) but the lamps on the phone are not blinking even if I setup one function key on my phones as shared line with number:

Re: [asterisk-users] Asterisk hangup all incoming calls after 10 seconds

Hi Bruno, I remember one of our customer had a similar problem with tellfree in Brazil. Their IT technician told me it was due to a g729 codec problem...they installed it and the problem disappeared. I never checked, I could only trust their man. Maybe it can help. Giorgio P.S.: let me know

[asterisk-users] mysterious rippled sound with IAX

Hi all, I know this could sound like a ghost story but... I have prepared an Asterisk 1.4.26.2 PBX on a Debian Lenny, the same I always prepare, same hw, same sw. I connected it to 2 iax phones using a hub: it works! I take everything it to our customer place, same PBX, same phones, same hub,

[asterisk-users] asterisk billing transferred calls

Hi, I'm looking for an application to show all the calls received/made including (this is very important!) transferred calls because I need to track all the time spent on the phone by all my employees. There is a list here but they are too many to try them all:

Re: [asterisk-users] anonymous calls code

Hi C F, I forgot to say it does not work with all telcossome telcos want *67#X in the DIAL string and some others want the key pad element method. Giorgio Incantalupo C F wrote: Huge thanks for mentioning what type of channel you are using. On Tue, Dec 22, 2009 at 5:11 AM, Giorgio

Re: [asterisk-users] anonymous calls code

[2-9]XX,1,SetCallerPres(prohib) exten = _*67[2-9]XX,2,Dial(Zap/g1/${EXTEN:3}) On Mon, Dec 21, 2009 at 6:19 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: Hi all, does anybody know how to make on-demand anonymous calls? I've tried code *67# before the number

Re: [asterisk-users] anonymous calls code

Hi C F, I solved the problem!! It was under my nose... If you are interested the solution is here: http://www.misdn.org/index.php/FAQ_chan_mISDN The right section is: key pad elements Giorgio Incantalupo C F wrote: You would have to create a dialplan for it. If your provider expects *67

[asterisk-users] anonymous calls code

Hi all, does anybody know how to make on-demand anonymous calls? I've tried code *67# before the number to call but it is working with some providers only. Any hints? Thank you. ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Asterisk crashes : Failed to start PBX

Hi Neo, have you checked your log files? It sometimes happened to me that Asterisk crashed without a reason. I discovered my logrotate didn't make its dirty work so I had huge log files. I lowered Asterisk log level and forced logrotate to work and now I have no more crashes. Hope it may

[asterisk-users] asterisk crashes when calling gtalk user

Is anybody experiencing the same? Found any workaround? Thank you. Giorgio Incantalupo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] calls drop during attended transfer with PRI line

it? I can send you the telco log if you need it. Thank you. Giorgio Incantalupo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk

[asterisk-users] asterisk and openvpn and sip

Hi all, I'm trying to connect one phone to a remote asterisk server via openvpn. First of all, I put the vpn server on the box hosting asterisk and the vpn client on another box, both with public ips. Then I set the client ip as my phone IP gateway and the remote pbx ip as the registrar and

Re: [asterisk-users] asterisk and openvpn and sip

, Giorgio Incantalupo wrote: Hi all, I'm trying to connect one phone to a remote asterisk server via openvpn. First of all, I put the vpn server on the box hosting asterisk and the vpn client on another box, both with public ips. Then I set the client ip as my phone IP gateway and the remote

Re: [asterisk-users] asterisk and openvpn and sip

=no' in your sip.conf entry for this phone? If not, you should. On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote: Hi John, I already have the ccd dir with the iroute (mandatory for routing to pc/phone connected to vpn client). During the last test I could register and make a call but voice

Re: [asterisk-users] problem with transfer application (REFER)

procedure / configuration to use a complete and stable implementation of the REFER functionality? Thanks to all in advance -- Giorgio Incantalupo, mailto:gincantal...@fgasoftware.com vo...@work - The Agile PBX http://www.voiceatwork.eu FGA srl - http://www.fgasoftware.com Tel: 02 997663.14

Re: [asterisk-users] rasterisk r processes take the rest of my cpu

Hi all, If anyone interested, it is a DUNDI bug which makes Asterisk crash with a segfault: disabling dundi fix the problem. Giorgio Incantalupo wrote: Hi all, I was playing with top on my Asterisk 1.4.24 server when I noticed this strange thing: PID USER PR NI VIRT RES SHR S

[asterisk-users] rasterisk r processes take the rest of my cpu

Hi all, I was playing with top on my Asterisk 1.4.24 server when I noticed this strange thing: PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 26797 asterisk 25 0 70524 14m 6416 S 1.3 2.9 5:59.44 asterisk ... 26518 asterisk 25 0 3316 1452 1140 R 46.6 0.3

[asterisk-users] activate telco redirection service from Asterisk

Hi, I want my telco to redirect all the incoming calls to my Asterisk towards another number (connected to my old Panasonic PBX) so I can stop Asterisk and repair my office. I tried to send the code *#21# ( Dial(mISDN/1/*#21#) ) but I get a busy channel while it is working with an ISDN

Re: [asterisk-users] activate telco redirection service from Asterisk

-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio Incantalupo Sent: Thursday, April 02, 2009 9:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] activate telco redirection service from Asterisk Hi, I want my telco

[asterisk-users] url in dial command: how does it work?

Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio

Re: [asterisk-users] url in dial command: how does it work?

Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo

Re: [asterisk-users] url in dial command: how does it work?

Hi Tim, it seems that using trunks is the right wayis this what you meant? Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make

Re: [asterisk-users] url in dial command: how does it work?

this. What were you planning to do with it. Tim. On 16 Mar 2009, at 13:04, Giorgio Incantalupo wrote: Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote

Re: [asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

Hi Gavin, if you can make and receive calls it works...do not worry if your line is shown as DOWN, some telco turns it off but it works without problem. Remember to ask your telco for the right signalling and set it the right way (PTP or PMP). Giorgio Incantalupo Gavin Henry wrote: Hi All

[asterisk-users] tons of open SIP channel between two snom 360

Hi, I'm monitoring an Asterisk 1.2.18 box because sometimes I get two Snom 360 phones creating a lot of SIP channels between them and it seems they never die. How can it be? Thank you. Giorgio A show channels excerpt follows: SIP/20-08a7aa80 (None) Up Bridged

[asterisk-users] pci cards VS patton

Hi all, I've been using telephone cards for 4-5 years and now I'm considering the Patton gateways. What are the pro and cons of Patton stuff compared to internal cards in a production system? Someone say external gateway are better because has no echo but have a longer delay when placing

Re: [asterisk-users] pci cards VS patton

Hi Olivier, so if you say that Patton hardware is bad documented, hard to configured and without echo canceller I think it is useless...don't you think? Unless it is much more reliable (no crashes at all) Olivier wrote: 2009/3/2 Giorgio Incantalupo gincantal...@fgasoftware.com

Re: [asterisk-users] asterisk and DNS

/asterisk-users -- _ Giorgio Incantalupo, mailto:gincantal...@fgasoftware.com FGA srl - http://www.fgasoftware.com - vo...@work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

[asterisk-users] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call

Hi, just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same zaptel/libpri/mISDN/add-ons. It crashes when transferring a call. Anybody tried it with success? Thank you Giorgio ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] T.38

Hi Michel, it seems there is a codec translation in between, have you tried to avoid it setting the codec from g729 to ulaw? I personally make Asterisk use alaw/ulaw codecs when sending faxes without any kind of codec translation and it seems to work. Giorgio michel freiha wrote: Dear All,

Re: [asterisk-users] WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to 'unknown sip:[EMAIL PROTECTED]

Hi Aldo, sorry not having posted it. ::)) You have to set canreinvite=no. Giorgio Incantalupo Aldo Alexander Leyva Alvarado wrote: What parameter??? 2008/1/17 gincantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi Olle, that was a phone misconfigurationa parameter

Re: [asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz

Hi JR, Tried with another motherboard: it works!! Tried with Asterisk 1.2 and 1.4: it works!! I cannot still believeit works...IT WORKS!!! Need to make some other tests with other kernels/machines/cards but I'm sure this is the right way! Thank you thank you thank you!!! ::-)) Giorgio.

[asterisk-users] ztdummy: rtc: lost some interrupts at 1024Hz.

. -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- Bandwidth

Re: [asterisk-users] Asterisk 1.4 or 1.6 ???

Hi, I agree with Gordon. We are still using Asterisk 1.2 because we are waiting for Asterisk 1.4 features to work as for Asterisk 1.2 (it seems to us that parking and queues have some problems... so not good enough for production). Giorgio Incantalupo Gordon Henderson wrote: On Mon, 6 Oct

Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4

Hi Mark, made some other tests but the problem remains. I installed 1.4.22-rc5 but nothing changed. I opened an issue on mantis waiting for a fix. Giorgio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 -

Re: [asterisk-users] How can Block a pri channel

-- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- Bandwidth and Colocation

[asterisk-users] setting DNID

Hi, I'm using Asterisk 1.2. I have to redirect a call coming from a line with DIDs to an ATA devices but keeping the DNID just as Asterisk would be DNID-transparent. I need this because the machine connected to my ATA needs to know which DID was called from outside. Anybody knows if DNID can be

Re: [asterisk-users] Fax with asterisk

Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable, use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote: Hi! I'm new

Re: [asterisk-users] Fax with asterisk

Hi Olivier, We DO NOT use faxdetect because it does not work properly. That's why we link a PRI DID to it, so when people call that DID the fax machine gets direct fax data without passing thru faxdetection. Giorgio Incantalupo. Olivier wrote: 2008/9/23 Giorgio Incantalupo [EMAIL PROTECTED

Re: [asterisk-users] Fax with asterisk

Hi Erik, once we used grandstream ATAs but now we are using linksys models: it has better design (look is important too for customers) and has 2 ports for two analog devices. We tested it with PRI and BRI lines and it seems working fine! Giorgio Incantalupo Erik Haider Forsen wrote: Hi

[asterisk-users] dundi and zap devices

Hi, is it possible to use zap devices (es: old analog phones) with dundi? It seems that a sort of zapregistration like sipregistration and iaxregistrations to include in the extensions.conf is missing... If yeshow? Thank you. Giorgio. ___ --

Re: [asterisk-users] how to force Asterisk 1.4 to use soxmix

Hi Julien, the soxmix (or sox in Asterisk 1.4 as default choice) is used by Asterisk to record queues calls when you ask it to mix the in and out calls, so I do not have control on it. Asterisk 1.2.x uses soxmix while Asterisk 1.4.x uses sox instead but the command sox launched by Asterisk

[asterisk-users] how to force Asterisk 1.4 to use soxmix

Hi, is there anybody who knows how to force Asterisk 1.4 to use soxmix instead of sox? Thank you. Giorgio ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:

Re: [asterisk-users] Dundi Help

tried to do it myself so I generated the two keys (pri and pub) for each server with their own hostname then I copied: - .121 keys to the other two servers (.137 and .204) - .137 keys to .121 - .204 keys to .121 Let me know how if it works. Giorgio Incantalupo technocrat voip wrote: Hello All

Re: [asterisk-users] Dundi Help

technocrat voip wrote: hi gior, If i understand correctly your setup would be like below. A is the dundi serer B is one pbx C is one pbx yes B and C dundi.conf contain the entity detials of A. yes Either for C or B we can place calls to the extensions registered on the other server.

Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4

Hi Mark, it seems that upgrading to Asterisk 1.4.21.2 magically solved the problem. :-) Giorgio Incantalupo Mark Michelson wrote: Giorgio Incantalupo wrote: Hi Mark, I assure my queues.conf is full of autopause = no, in the singles and general contexts (I'm not sure where to put

Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4

Michelson wrote: Giorgio Incantalupo wrote: Hi Mark, it is show queues I use to see if phones are paused or not. The phones I'm using for tests are all SIP phones. Yes, what you are supposing could be right...Asterisk could see the phones as stuck. I'm still investigating, making test on my

[asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4

Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes Anybody knows why a phone becomes paused? Is it an

Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4

!?!? Giorgio Mark Michelson wrote: Giorgio Incantalupo wrote: Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf

Re: [asterisk-users] Zaptel problem with pots lines

Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http

Re: [asterisk-users] Asterisk unable to register to tnet.it

from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it

Re: [asterisk-users] Asterisk unable to register to tnet.it

detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without

Re: [asterisk-users] Asterisk unable to register to tnet.it

is not correct :-) Do you already test to just ping to tnet.it port 5060 ? Marino On Tue, Jul 15, 2008 at 10:27 AM, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Marino, 1) yes I can connect using the account 2) no, I'm running zoiper on a different machine. I'm using an Asterisk server

Re: [asterisk-users] language problem

/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

[asterisk-users] Asterisk unable to register to tnet.it

Hi, I cannot make my Asterisk register to tnet.it, an italian SIP provider. I tried many register string formats and tried to set realm and outboundproxy (sip.tnet.it) too but without any result. Still I cannot register (but for example messagenet works fine). Is there anybody who tried this

Re: [asterisk-users] Asterisk unable to register to tnet.it

a look at SIP connection messages from and to this SIP server? I suggest you to use wireshark to check sip messages. Thanks, Marino On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I cannot make my Asterisk register

Re: [asterisk-users] changing inbuilt sound messages

/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

[asterisk-users] Asterisk hangup not working on inbound calls

? Thank you. Giorgio Incantalupo. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] cell phone hangup not getting recognised by system

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http

[asterisk-users] Digium stopped TDM400P production: alternatives??

Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a fully-compatible Openvox product...but is it really fully-compatible? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio.

Re: [asterisk-users] Asterisk and fax

Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote: Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one

Re: [asterisk-users] Suggested BRI cards?

-- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] modprobe

Hi Josu, I had the same problem with wctdm.I just loaded zaptel before wctdm and it was all ok. Hope it can help you. :) Giorgio Incantalupo Josu Lazkano wrote: Hello every boy again I have some problems with modprobe. When I type modprobe zaphfc, this error happens FATAL: Module

[asterisk-users] mISDN: long delay when making outbound calls

searched on misdn.org but found nothing. I'd like to understand if this delay is caused by telco line or the driver and in the latter case if there is a way to shorten this delay. TIA Giorgio Incantalupo -- _ Giorgio Incantalupo, mailto:[EMAIL

[asterisk-users] module zttranscode: what is it?

Hi, does anybody know what *zttranscode *module* *is for*?* Thanks!! Giorgio -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14

[asterisk-users] asterisk slows down when unplugging internet cable with VoIP lines

this behaviour or should I avoid VoIP lines?? 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

Re: [asterisk-users] asterisk slows down when unplugging internet cable with VoIP lines

1) or clear your DNS cache and force a lookup again (option 2). However, most providers won't change the IP on your, and I would hope that if they did, they would notify their customers ahead of time. Cheers, Alex Robar On 4/26/07, *Giorgio Incantalupo* [EMAIL PROTECTED] mailto:[EMAIL

[asterisk-users] Asterisk+mISDN drops calls after 3-4 secs

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after

Re: [asterisk-users] Asterisk+mISDN drops calls after 3-4 secs

Hi all, problem solved! It was a telco problem. Giorgio Incantalupo Giorgio Incantalupo wrote: Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version

Re: [asterisk-users] Asterisk 1.2.16 - No Caller ID

Hi, have you tried different values of callerid? Maybe setting *useincomingcalleridonzaptransfer* to yes can help you. Giorgio Incantalupo OCOSA ListAcc wrote: Hello, When I upgraded a while back the caller ID stop working I have tried everything and searched the lists no answer. Please

Re: [asterisk-users] asterisk unable to create files, too many files open

Hi Maysara, I have your same problem. are you using mISDN? If yes update your driver. Giorgio Incantalupo Maysara A. Abdulhaq wrote: hello, im having trouble with asterisk with medium load, it seems im running out of files, here is a chunk of the logs with grep \(file\|pipe\): Apr 18 15

Re: [asterisk-users] Delay to start sip registration after asterisk restart

Hi Frederico, I sometimes have the same problem tooI think the problem is related to VoIP providers registrations. Are you using VoIP services on your PBX? Thank you. Giorgio Incantalupo Frederico Madeira wrote: Hi, My asterisk was working fine but today my calls won't out of my

[asterisk-users] SNOM and Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip

Hi, I have an Asterisk 1.2.9.1 box and a bunch of snom phones. I sometimes get this error: *ERROR[31201] chan_sip.c: Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip* It seems that my SIP phone is sending subscribe command for numbers not inserted inside

Re: [asterisk-users] Couldn't load variables.txt?aldope=xxxxx

Hi Carlos, I suggest you to use /init/op_panel_debian.sh script inside oppanel tar file. Put it inside /etc/init.d and then as root type: *update-rc.d op_panel defaults * to setup the script for boot. Giorgio Incantalupo Carlos Jerónimo wrote: Hi Giorgi thanks for all, it works. Your

Re: [asterisk-users] SNOM 360

Hi UxBoD, just create a voicemail for your extension and Asterisk will do the rest!!! Giorgio Incantalupo --[ UxBoD ]-- wrote: Hi, I have got my new phone working with Asterisk, and must say it is very very good combination. Now I have WMI working, but what I would like to be able to do

Re: [asterisk-users] GTalk/Jabber passing audio in 1.4.1!

Hi Ronald, I can make my gtalk client connect with my Asterisk 1.4.1 infact jabber show connected CLI command shows one user connected. When I call I get no voice: I suspect it is from gtalk server to my Asterisk because if I send a voice mail via gtalk I hear the female voice who tells me to

Re: [asterisk-users] Couldn't load variables.txt?aldope=xxxxx

Hi Carlos, type: *ps -A -F | grep panel* You should see something like: root 14851 1 0 2700 8164 0 11:01 ?00:00:01 /usr/bin/perl -w /usr/local/op_panel/op_server.pl -d -p /var/run/asterisk/op_panel.pid This means that tha panel process is running. Giorgio Incantalupo

Re: [asterisk-users] sip: failed the authenticate on INVITE

Hi Michael, have you tried to set canreinvite = no inside incoming calls context in sip.conf? Some SIP provider does not like reinvite. Giorgio Incantalupo Michael Zoller wrote: I've got a problem with a SIP Account I am trying to dial in with. The correct extension rings but when I pick

Re: [asterisk-users] How to place a call to a Google Talk user?

Hi Am, I've got a similar problemyou mean you can connect and call but hear no sound?? Giorgio Am Turnip wrote: I am trying to dial a GTalk, ie @gmail.com, address. I inscribed this address in jabber.conf on the buddy= line. Upon executing the Dial application, I hear only a brief

Re: [asterisk-users] Couldn't load variables.txt?aldope=xxxxx

inside its directory) so you can see it is working (you'll see a lot of messages). If you cannot find the oppanel dir this means it is not installed. You could download it from www.asternic.org and install it following the instructions on the site. Giorgio Incantalupo Carlos Jerónimo wrote

Re: [asterisk-users] Couldn't load variables.txt?aldope=xxxxx

Hi Carlos, this happens to me when oppanel server is not working. Check it is running. Giorgio Carlos Jerónimo wrote: HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get

[asterisk-users] * 1.4.1: connected to gtalk but no voice passing

Hi, I managed to connect Asterisk 1.4.1 to my gtalk account but after calling I hear no voice from other side (a SIP phone). Asterisk log says nothing. What am I missing? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided

Re: [asterisk-users] Can I generate random SIP traffic?

Hi Gabriele, maybe sipp can help you: http://sipp.sourceforge.net/ Giorgio [EMAIL PROTECTED] wrote: Hello, I would like to generate a peer-to-peer or a server/client SIP traffic between two or more Openwrt access point, to make some statistics about QoS. I tried some SIP traffic generators

Re: [asterisk-users] Moving from Bristuff to mISDN

Hi Olivier, ISDN channels are not seen as zap channels as with briStuff so you have to edit misdn.conf to use them. If you are using your own framework above Asterisk architecture you to make some changes to manage new type of channels. Giorgio Incantalupo Olivier wrote: Hi, Which

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