[Asterisk-Users] AGI commands STDOUT problem

2005-03-24 Thread Moises Silva
Reading 80 bytes response from Asterisk... Received response: agi_uniqueid: 659738.1 Reading 80 bytes response from Asterisk... Received response: agi_callerid: Moises Silva 12 Reading 80 bytes response from Asterisk... Received response: agi_dnid: 10 Reading 80 bytes response from

Re: [Asterisk-Users] VoIP network configuration using Asterisk and SIP

2005-04-05 Thread Moises Silva
Hi. I have no time to read the whole configuration of each device, but first, i guess you have to be sure that each PBX is able to reach the other networks, may be a simple ping can tell you this. After that, i think you have to make sure that no firewall is blocking the 5060 port for the SIP

Re: [Asterisk-Users] Context overlap?

2005-04-06 Thread Moises Silva
, if hopefully im understanding your question. Even tough they are in different context.. i think thats wrong, since you are including both contexts in auto-attendant, they both are in the same context. The bussinesshours run first because its included first (in the hours set in the

Re: [Asterisk-Users] Problem with fxo

2005-04-12 Thread Moises Silva
I have no Idea of the strange errors, but as far as i know, the proper way of calling is: Zap/g${group}/${phone_number} where ${group} is a valid group inside zapata.conf, and ${phone_number} is the desired PSTN phone to call. In you email you wrote the messages and i can see that you missed

Re: [Asterisk-Users] How to get list of codecs

2005-04-12 Thread Moises Silva
mmm i think Agi by itself does not provide a way to do so. And the codecs are negotiated depending upon the codec that both call sides support. So, i belive that the only way is making your own implementation of AGI in res_agi.c :) Hopefully someone will come up with a better idea :-) best

Re: [Asterisk-Users] ZAP channel hangs up with no apparent reason

2005-04-13 Thread Moises Silva
what is the configuration you have in zapata.conf ??? try using callprogess=no and busydetect=no, and if you have trouble to hangup the calls, then try callprogress=no and busydetect=yes ... i had troubles when both parameters were set to yes. Another usefull parameter is 'busycount', it

[Asterisk-Users] about volume in Playback() files

2005-04-15 Thread Moises Silva
Hi. How can i increment the volume of the files played with Playback or Background() ??? thanks in advance. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Asterisk connect to Asterisk

2005-04-16 Thread Moises Silva
Hi Lee. Actually is very easy to connect to asterisk servers. Please read the IAX related documentation in voip-info: http://www.voip-info.org/wiki-Asterisk+config+iax.conf and if you have troubles, ask here :-) Best Regards On 4/16/05, lee siang fong [EMAIL PROTECTED] wrote: Hi, I am

Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

2007-07-18 Thread Moises Silva
Alvaro, can you post the patch in a public place and post the URL here? It might be a good idea to contact steve underwood to see what he has to say about such a patch. Regards, On 7/18/07, Alvaro Parres [EMAIL PROTECTED] wrote: Carlos: Only for check do this change:

Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

2007-07-23 Thread Moises Silva
Alvaro, Naming Asterisk versions is of little help since Asterisk is not the one failing here. It would be more helpful know the libmfcr2 and spandsp versions that were used in the working/non working tests, is that possible? do you have the versions at hand? Thanks a lot. On 7/23/07,

Re: [asterisk-users] Unicall and Private CID

2007-08-03 Thread Moises Silva
Carlos, If you are interested we can meet us via MSN someday to debug the problem. I don't know if that's possible though, since it seems is your production server. Moy On 8/2/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Hi Carlos, I suggest you download spandsp-0.0.3pre22.

Re: [asterisk-users] Unicall and Private CID

2007-08-03 Thread Moises Silva
I would not call that properly a fix. We need to know why is failing in newer spandsp versions in the first place. Can you make a diff and post it? On 8/3/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Fri, 2007-08-03 at 00:23 -0300, Luis Antonio Prata Barbosa wrote: Hi Carlos, I suggest

Re: [asterisk-users] Unicall and Private CID

2007-08-04 Thread Moises Silva
On 8/4/07, Steve Underwood [EMAIL PROTECTED] wrote: Why are people so determined to break things. If you want to use unicall-0.0.3pre11, use it with spandsp-0.0.2. Not really determined to break things, but to understand failures, even when those failures are because of version missmatching :)

Re: [asterisk-users] Problem in installing libmfcr2 for configuring MFC/R2

2007-08-17 Thread Moises Silva
I think you did not installed properly the libraries. libmfcr2 check for: checking tiffio.h usability... yes checking tiffio.h presence... yes checking for tiffio.h... yes checking for TIFFOpen in -ltiff... yes That is, you need to have BOTH, headers and shared library. ldd in protocol_mfcr2.so

Re: [asterisk-users] unable to load chan_unicall.so

2007-08-23 Thread Moises Silva
Edit logger.conf and learn how to enable debugging,verbose and all kind of messages. Enable all levels of messages, try again and tell us what is the error message exactly. Regards, On 8/23/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am using debian 4.0 with version

Re: [asterisk-users] Error in loading libunicall.so module while running asterisk command

2007-08-24 Thread Moises Silva
If you still have this problem, contact me via MSN at the same address I write from. Im sure that with 5 minutes in your box we can fix it. Regards On 8/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On 8/23/07, [EMAIL PROTECTED] Hi, I am using debian 4.0 with version

Re: [asterisk-users] app_conference

2007-08-30 Thread Moises Silva
Anton, I used app_conference last year, debugged some problems with voice frames of 240 samples and made some fixes to the code. This is the result: http://www.moythreads.com/app-conference-ast-1.2.12.1-nov-6-2006.tar.bz2 I reported the problem to iaxclient-devel mailing list, as noted here:

Re: [asterisk-users] app_conference

2007-09-02 Thread Moises Silva
, Anton Krall [EMAIL PROTECTED] wrote: Hi Moises. So, would you recommend app_conference over meetme? Knowing what you know about it? Saludos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: jueves, 30 de agosto de 2007 09:06

Re: [asterisk-users] Manager Originate without phone off hook?

2007-09-03 Thread Moises Silva
May be I am missing something, but, manager command DBPut should do the trick of putting the DB value. And, since you are already using the manager interface, you are using PHP or PERL to connect to the Database, why not wait for the DBPut command response and from the script execute wget??

Re: [asterisk-users] Asterisk Died message

2007-09-04 Thread Moises Silva
Signal 11 is a segmentation fault, if you are not running unsupported patches on Asterisk you should compile without Asterisk optimizations and open a bug attaching the debugging backtrace. Read This: http://www.voip-info.org/wiki-Asterisk+debugging Regards, On 9/4/07, Nitesh Divecha [EMAIL

Re: [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault

2007-09-11 Thread Moises Silva
Open a bug in http://bugs.digium.com/ including all the information you provided here. Also remember to read the bugs guidelines before openning the bug, this might be already reported. Regards On 9/11/07, Bruce McAlister [EMAIL PROTECTED] wrote: Hi All, I have a really strange issue

Re: [asterisk-users] Using PHP to reload extensions

2007-10-03 Thread Moises Silva
If you are running the script from a web server, the script gets executed with the web server process permissions, hence, probably does not have access to /var/run/asterisk.ctl. You can give permissions to your web server, or better yet, dont execute the command using shell_exec, better open a

Re: [asterisk-users] A Leg Control on Asterisk Callback

2007-10-29 Thread Moises Silva
Why dont you make 2 separate Originate actions, one for each call leg. Then call Bridge manager Action whenever you want. Moy On 10/29/07, Douglas Garstang [EMAIL PROTECTED] wrote: I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an

Re: [asterisk-users] MFC/R2 on AsteriskNOw

2007-10-29 Thread Moises Silva
just install chan_unicall.so On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: MFC/R2 on AsteriskNOw!! How? Please!!! Thanks!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To

Re: [Asterisk-Users] SIP, NAT and Firewalls

2006-01-20 Thread Moises Silva
you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michaël Gaudette [EMAIL PROTECTED] wrote: Hello, I'm a bit new to SIP, and I've set up a SIP

Re: [Asterisk-Users] Music on Hold

2006-01-20 Thread Moises Silva
with so poor explanation of your problem i hope your asking payed help. regards On 1/20/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I tried everything to get the music on hold feature to work with [EMAIL PROTECTED], to no avail. Need some help to get it running. Thanks, Eduardo

Re: [Asterisk-Users] SIP problem picking up the call

2006-01-21 Thread Moises Silva
sip debug rtp debug enable all the log levels in console in logger.conf regards On 1/20/06, RumaTech [EMAIL PROTECTED] wrote: Hi, all I am trying to call to particular destination via SIPNET (one of the VoIP providers). I can succesfully dial and I can hear waiting tone, however nothing

Re: [Asterisk-Users] No congestion

2006-01-21 Thread Moises Silva
check incominglimit and outgoinglimit in sip.conf http://www.voip-info.org On 1/20/06, Kristian Larsson [EMAIL PROTECTED] wrote: Hey! I'm having a small problem. I'm using Realtime to store SIP account information. Dialing works just fine, but when dialing a person already on the phone I

Re: [Asterisk-Users] Bug in attended transfer or as expected?

2006-01-23 Thread Moises Silva
The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. If you have enabled Disconnect Call feature, then you can hangup with *0 for example, that will hangup only the current

Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(

2006-01-23 Thread Moises Silva
please turn on all the debug, warning, error etc messages in the console, see logger.conf, then type sip peer peer1 debug and sip peer peer2 debug to see the SIP messages. How are you testing if asterisk is in the media path? Regards On 1/23/06, Steve Gladden [EMAIL PROTECTED] wrote: been

Re: [Asterisk-Users] Bug in attended transfer or as expected?

2006-01-25 Thread Moises Silva
Its not so hard to look into the source code and make small changes. Im not sure how hard it would be to implement what you want, but i have tested it, and yes, you are right, the call get disconnected, and i agree that souldnt be that way. You may want to open a feature request in bugs.digium.com

Re: [Asterisk-Users] Re: SIP RTP Negotiation

2006-01-25 Thread Moises Silva
Few people, or no one, will take the time to see all the debug. The key here is that the RTP port and IP negotiated in the SDP message sent by asterisk to each party, should be visible for the party. A common error is Asterisk sending in SDP a private IP address to a public UA, so the public UA

Re: [Asterisk-Users] Preventing Asterisk from transfering the call

2006-01-31 Thread Moises Silva
well. Im supposing you mean a SIP phone. Transfers with SIP phones happens to be a method called REFERRER. Im not sure if its a feature of Asterisk to allow the administrator to ban the referrers, but if is not a feature, letme know, may be i can make a patch soon. To look for a feature like

Re: [Asterisk-Users] Help configuring Asterisk server

2006-01-31 Thread Moises Silva
please consider posting this as a Job offer in asteriskhelpdesk, because of your lack of information i can tell you are really stuck :DOn 1/30/06, Naren Koka [EMAIL PROTECTED] wrote:I need to configure / migrate Asterisk server from 0.9 to the latest version with some upgrades. Please help!Thank

Re: [Asterisk-Users] Regarding cdr_manager.conf

2006-02-02 Thread Moises Silva
Hi Victor. in /etc/asterisk/modules.conf you MUST have autoload=yes, or better yet, just load what you need INCLUDING the modules cdr_manager.so, you can test if you have it by doing in the asterisk console show modules, if you dont have it, you can load it immediatly from the console doing load

Re: [Asterisk-Users] return code from AGI

2006-02-03 Thread Moises Silva
not sure what you want, but for multiple returns i use Set(AGI_STATUS=mystatus), so in the dialplan i just check for the variable AGI_STATUS and do whatever i need depending on the status. regardsOn 2/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:Hello friends,Asterisk applications like Dial

Re: [Asterisk-Users] Fast AGI performance question

2006-02-04 Thread Moises Silva
Good question, i would like to know the same. Im using MAGI patch to execute AGI commands via the Manager. I have a PHP proxy connected to the CallManager PHP server that do the routing stuff and decide to execute Dial, Voicemail, Playtones, receive DTMF or some other stuff in the channel, i have

Re: [Asterisk-Users] attended call transfer

2006-02-09 Thread Moises Silva
this is a Normal behaviour, nevertheless i dont think is a correct behaviour. Several weeks ago other user asked the same, i suggested him to open a feature request on bugs.digium.com, check for that regardsOn 2/9/06, Thomas Artner [EMAIL PROTECTED] wrote: Hi!I am new with asterisk and I have my

Re: [Asterisk-Users] How can I send DTMF from the console?

2006-02-10 Thread Moises Silva
i dont think you can do it from the console unless you hack the code to be able to use http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF from the console. RegardsOn 2/9/06, Anthony Azzopardi [EMAIL PROTECTED] wrote: How can I send DTMF from the

Re: [Asterisk-Users] attended call transfer

2006-02-10 Thread Moises Silva
Yep, im interested in coding to solve that problem, unfortunately i havent had time. I hope to be free in 2 weeks and start looking in the code to see if i can do something. Unless some one else has done it already. Regards.On 2/10/06, Alex Barnes [EMAIL PROTECTED] wrote: -Original

Re: [Asterisk-Users] Dial command to connect two channels and bypassasterisk server

2006-02-14 Thread Moises Silva
Unless you use SIP ALG (Application Layer Gateway) like the module in netfilter to set the expectations? correct? RegardsOn 2/14/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Wai Wu wrote: If Asterisk is in the public network, it will work. The problem is when Asterisk is behind NAT and one of

Re: [Asterisk-Users] asterisk silence suppression?

2006-02-15 Thread Moises Silva
Asterisk DOES NOT HAVE silence suppression (VAD) support for now. So it cannot be disabled or enabled. Simply does not exists. A couple of weeks ago i saw a patch to enable it. The link here: http://bugs.digium.com/view.php?id=5374 so unless you have the previous patch, you should disable

Re: [Asterisk-Users] asterisk silence suppression?

2006-02-17 Thread Moises Silva
The patch you saw is not for the stable branch. Salu2 Jsalas Right, but try using this, i adapted it, no guarantees, i have not made tests, just modified it to apply properly, it would be great if some one can test it: http://chewbacca.ivsol.net/asterisk-1.2.1-silence-suppression-4.patch

Re: [Asterisk-Users] Keep getting message in logs that pbx.c cannot find extension context 'default'

2006-02-24 Thread Moises Silva
do you have a defaultcontext=something parameter in sip.conf [general] section?? If not, the default is... em default RegardsOn 2/23/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi,I am getting repeated messages in my logs with the following:*Feb 23 07:56:11 NOTICE[2470] pbx.c:

Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

2006-02-24 Thread Moises Silva
you need to set a TRANSFER_CONTEXT, either for the transferer or transferee channel. I dont know why, but res_features give priority to the transferee TRANSFER_CONTEXT, if not found, then use the transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this

Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

2006-02-24 Thread Moises Silva
it seems im not undestanding your question then. Could you provide a practical example?On 2/24/06, Chuck Bunn [EMAIL PROTECTED] wrote:Hi,Okay but then how do you transfer across contexts then? ThanksMoises Silva wrote: you need to set a TRANSFER_CONTEXT, either for the transferer or transferee

Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

2006-02-25 Thread Moises Silva
andresturant as well. ThanksMoises Silva wrote: it seems im not undestanding your question then. Could you provide a practical example? On 2/24/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Okay but then how do you transfer across contexts then? Thanks Moises Silva wrote: you

Re: [Asterisk-Users] Problems dialing to another Asterisk server

2006-02-27 Thread Moises Silva
At first sight there is no problem, your code looks good, no warnings etc, is just that nobody picks up in the other end. Do you have access to the other Asterisk server? what does the console shows up? I have not used the manager with Java, but that does not seems to be your problem. I guess you

Re: [Asterisk-Users] Asterisk transfer conflict

2006-03-01 Thread Moises Silva
see /etc/asterisk/features.conf On 3/1/06, Fredrik Jensen [EMAIL PROTECTED] wrote: I have a problem with my Asterisk system.When I use my phone to call my office mailbox I have to end my password with#. (The office do not use Asterisk) # is also used as a transfer button on my asterisk, so when I

Re: [Asterisk-Users] Preferred editor(s) dialplan coding?

2006-03-03 Thread Moises Silva
vim ;)On 3/3/06, Bill Gibbs [EMAIL PROTECTED] wrote: Vim for everything-Original Message-From: [EMAIL PROTECTED][mailto: [EMAIL PROTECTED]] On Behalf Of PeteBarnwellSent: Friday, March 03, 2006 7:39 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]

Re: [asterisk-users] Unicall mfcr2 testcall issues in mexico outgoing:ok | incoming: fail.

2008-02-28 Thread Moises Silva
This may fix your issue: mx,10,4,0 By default Mexico variant has the option get ANI after DNIS. Which it means just after getting the DNIS digits we will request the calling party category and DNIS. The Nortel PBX seems to not like calling party category requests and they want to go straight to

Re: [asterisk-users] DTMFR2- UNICALL

2008-03-06 Thread Moises Silva
What kind of problems are you talking about and what you want to modify? On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada [EMAIL PROTECTED] wrote: Hi Asterisk-user, Steve; I´m using libmfcr2-0.0.3.tar.gz,

Re: [asterisk-users] DTMFR2- UNICALL

2008-03-07 Thread Moises Silva
On 3/6/08, Moises Silva [EMAIL PROTECTED] wrote: What kind of problems are you talking about and what you want to modify? On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada [EMAIL PROTECTED] wrote: Hi Asterisk-user, Steve; I´m using libmfcr2-0.0.3.tar.gz

Re: [asterisk-users] Unicall + incomplete DNIS on international calls

2008-04-04 Thread Moises Silva
Hello Ivan, I don't see nothing wrong in terms of signaling. When your side (Asterisk/Unicall) request ANI, the other end answer with the signal F, which means No More ANI, hence you receive an empty ANI string. When your side request DNIS, the other end does not answer in several seconds, which

Re: [asterisk-users] G729 license count...

2008-04-17 Thread Moises Silva
http://store.digium.com/productview.php?product_code=G729CODEC http://www.digium.com/en/docs/G729/g729policy.php http://www.voip-info.org/wiki-Asterisk+G.729+Licensing On Thu, Apr 17, 2008 at 11:14 AM, Carlos Chavez [EMAIL PROTECTED] wrote: I need a refresher course on how many licenses

Re: [asterisk-users] Problems with Quality Voice in a Asterisk-E1-Unicall

2008-04-21 Thread Moises Silva
The E1 use ALAW, if you want to avoid trans-coding use ALAW in your phones as well. In any call you have 2 call legs, callee and caller, try to isolate the problem and determine if the audio is really coming that bad from the E1, you can use ztmonitor to hook into the E1 and listen to the audio.

[asterisk-users] MFC/R2 in chan_zap , Testers Wanted

2008-04-24 Thread Moises Silva
If you are an MFC/R2 user and want to help in the development of chan_zap support for this signalling, please take a look at the bugtracker at http://bugs.digium.com/view.php?id=12509 and/or contact me. Currently just México support is built-in, if you want your country variant supported, drop me

Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted

2008-04-24 Thread Moises Silva
Hello Moisés, thanks for your effort on this! I would love to use Digium cards for MFC/R2 signalling in the future. Currently you can use Digium cards with Unicall :-) , tho, having MFC/R2 on chan_zap is more handy. I added some info you might like in the bugtracker, you might take a look

Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted

2008-04-24 Thread Moises Silva
. Moisés Silva On Thu, Apr 24, 2008 at 8:26 AM, Ruben Zamora [EMAIL PROTECTED] wrote: Moises Thats means, that we arent going to use unicall? If that true i can test these weekend with a E1-Axtel. Thanks Ruben Moises Silva escribió: If you are an MFC/R2 user and want to help

Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted

2008-04-24 Thread Moises Silva
Way more handy and will be much more reliable too. Steve Underwood did a great job implemeting it, but as far as I know the code isn't actively maintained anymore. Of course your implementation of MFC/R2 will take a while to become stable, but hey -- it's a start. Agreed. Russel pointed

Re: [asterisk-users] MFC/R2 in chan_zap , Testers Wanted

2008-04-24 Thread Moises Silva
Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport of the Steve driver (now coded for Callweaver derivative) to Asterisk (1.2, 1.4, and 1.6 soon). It works pretty well. In fact, it works more stable in 1.4 than the original Steve driver in 1.2, and with better sound

Re: [asterisk-users] Unicall - How to automatically block collect calls

2008-05-05 Thread Moises Silva
The latest version of the driver included in http://www.moythreads.com/astunicall/ comes with a change that will set the variable UC_CATEGORY in your dialplan, Brasil has a special category for those calls, don't remember the name that will show up, but you can make a couple of tests and then drop

Re: [asterisk-users] Help-ASTERISK-MFCR2

2008-06-07 Thread Moises Silva
You need to enable loglevel=255 in unicall.conf and enable all the levels of logging in logger.conf, otherwise the logs you post don't say much. Moisés Silva On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone [EMAIL PROTECTED] wrote: Dears, I have problem ASTERISK with PSTN SIEMENS EWSD (MFC

Re: [asterisk-users] Help with MFC/R2

2008-09-17 Thread Moises Silva
That means someone else has already open the zap device, most likely Asterisk. Just one application at a given time can open a zap device. You cannot run testcall and Asterisk at the same time unless you make sure they don't try to open the same channels. Moy On Wed, Sep 17, 2008 at 1:27 AM, Dae

Re: [asterisk-users] Help with MFC/R2

2008-09-17 Thread Moises Silva
But after rerunning the test, I only get the first log (w/o Far end replies.) Any help will be really appreciated! Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 8:33 AM To: Asterisk

Re: [asterisk-users] Help with MFC/R2

2008-09-18 Thread Moises Silva
... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked

Re: [asterisk-users] Help with MFC/R2

2008-09-18 Thread Moises Silva
The line is supposed be E1-MFC/R2 and works perfect with a Panasonic PBX, actually Exists any variant of MFC/R2? And how can I configure it to get working? As I said, no matter which variant you try, the AB bits MUST be in 10 to be able to make calls with Unicall/libmfcr2. I have never seen

Re: [asterisk-users] Help with MFC/R2

2008-09-19 Thread Moises Silva
] On Behalf Of Moises Silva Sent: Wednesday, September 17, 2008 10:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with MFC/R2 It seems to me your lines are blocked. Execute zttool and if you see 1101 in the rx bits, it means the telco

Re: [asterisk-users] Help with MFC/R2

2008-09-20 Thread Moises Silva
, How i can do to join asterisk-r2 list ? My congratulations about your article in digium blog http://blogs.digium.com/page/2/ I will collaborate in your project and give support from Venezuela. Regards, Luis Morales On Sat, Sep 20, 2008 at 7:47 PM, Moises Silva [EMAIL PROTECTED] wrote: Dae

Re: [asterisk-users] DTMF issues...

2008-10-03 Thread Moises Silva
Hey Carlos, What is the best method to debug DTMF issues? Do I have to sniff the SIP packets? The best method to debug DTMF issues depend on how you receive those DTMF digits. Assuming you can use SIP INFO for the DTMF, that means the DTMF digits are not really DTMF :-), that is, is

Re: [asterisk-users] OPENR2 in Thailand

2008-10-20 Thread Moises Silva
Hello Peter, You can ask this better in the asterisk-r2 mailing list. I don't know of anyone that has used OpenR2 in Thailand, but I am interested in adding support for that variant. Contact me at this same e-mail address or via Google talk (my e-mail address works for MSN as well ) to discuss

Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

2007-04-20 Thread Moises Silva
Thanks a lot for the fix Humberto. On 4/18/07, Humberto Figuera [EMAIL PROTECTED] wrote: Hi Moises, the Asterisk SVN-branch-1.4-r60989 make a change in the ast_channel_alloc function: This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as

Re: [asterisk-users] Unbridge the two call.

2007-05-10 Thread Moises Silva
Have you tried using Redirect, manager action? On 10 May 2007 11:47:16 -, pandi ponnangan [EMAIL PROTECTED] wrote: Hello all, I will bridge the twoexisting call using BRIDGE command.after that i want to unbridge the two call. For example 1) A and B in conversation(using BRIDGE

Re: [asterisk-users] Asterisk and unicall + mfcr2 signalling

2007-05-14 Thread Moises Silva
try using testcall with 255 as debug level and report back results in order to be able to help you. http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf On 5/14/07, Joca Loco [EMAIL PROTECTED] wrote: Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P

Re: [asterisk-users] Asterisk 1.4 with Unicall

2007-06-09 Thread Moises Silva
Hi Carlos, On 6/8/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a small call center running with Asterisk 1.4.4 and Unicall. Everything seems to be working but twice now we had to reset the server because all lines stopped working. You can see users dialing in and reaching the

Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

2007-06-09 Thread Moises Silva
Alvaro... Hum..., I never have tried RxFax... let me know if you need any extra help with that. Sounds interesting On 6/8/07, Alvaro Parres [EMAIL PROTECTED] wrote: Moy: I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem i have is the RxFAX application, that broke

Re: [asterisk-users] ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args??

2007-06-10 Thread Moises Silva
va_list is a macro used to define optional arguments, so I suppose is correct to call it with 5 or more arguments. I suggest you to compile asterisk without optimizations ( search in voip-info.org ) and then open a bug ( make sure first you read the bug guidelines ) in bugs.digium.com Regards

Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

2007-07-17 Thread Moises Silva
In order to help you I need testcall traces, with max level of logging, of incoming Nextel calls. Regards, On 7/17/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Tue, 2007-07-17 at 15:39 -0500, Victor Toofic wrote: El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba: I

Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian

2007-01-17 Thread Moises Silva
Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote:

Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian

2007-01-19 Thread Moises Silva
of timers only? Greets! On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2

Re: [asterisk-users] using the Manager to connect caller to conference

2007-01-19 Thread Moises Silva
see Originate manager Action in voip-info.org On 1/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Is there a way, via the Asterisk Manager, to dial an extension and connect it with an existing meetme conference? I'm trying to pull callers into a conference as other conference members

Re: [asterisk-users] chanskype

2007-01-20 Thread Moises Silva
Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error So you think is easy to us guess wich error you are getting? Seriously, I think you should read this: http://www.catb.org/~esr/faqs/smart-questions.html Anyone has experience with this?

[asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

2007-02-04 Thread Moises Silva
this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/ Kind Regards Moises Silva -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided

Re: [asterisk-users] Something wrong with the list?

2007-02-07 Thread Moises Silva
same for me, however today I started receiving the same amount as usual On 2/6/07, C F [EMAIL PROTECTED] wrote: Since Monday I didn't see much traffic. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

Re: [asterisk-users] phpagi - Event On Hangup

2007-02-12 Thread Moises Silva
Usually you should use the manager interface for that. On 2/12/07, nik600 [EMAIL PROTECTED] wrote: Do you know if it is possible to handle some events with phpagi? For example: On hangup (doesn't care if by caller or by asterisk) do something Thanks

Re: [asterisk-users] Playback uses channel's language, background doesn't

2007-02-26 Thread Moises Silva
it may be a bug, try creating a simple test script with only 2 extensions, one with playback the other one with background and see how it works, also post here the asterisk version you are using. Regards On 2/26/07, kjcsb [EMAIL PROTECTED] wrote: I have the following in the dialplan:

Re: [asterisk-users] Playback uses channel's language, background doesn't

2007-02-27 Thread Moises Silva
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage There you can found how you can get the current language ( the same used by playback ), so you can set a local variable to the current language and use it instead of the blank value Regards On 2/26/07, kjcsb [EMAIL

Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

2007-03-14 Thread Moises Silva
. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/ Kind Regards Moises Silva ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Problems with MFCR2 and Meridian

2007-03-16 Thread Moises Silva
Arturo, the error does not says much really, just that either the other end timed out expecting a response from you, or your end timed out expecting a response from the other end :) However, from my experience, it may be an error in your DNIS/ANI configuration and/or an mfcr2 library error (

Re: [asterisk-users] make Zaptel 1.2.16 'struct inode' has no member named 'u'.

2007-04-04 Thread Moises Silva
Zaptel has no direct code relationship with Asterisk. Your error is because zaptel is trying to use a member no longer exists in newer kernels. However you are using fedora, and fedora included that change in older kernel. I found this in xpp/xbus-core.c /* * As part of the inode diet the

Re: [asterisk-users] Manager Originate and Var to long

2007-04-08 Thread Moises Silva
you can easily increment the buffer size changing include/asterisk/manager.h #define AST_MAX_MANHEADER_LEN 256 chage that line for something like this #define AST_MAX_MANHEADER_LEN 512 and recompile Asterisk, Is the only way I know, Regards On 4/8/07, Thomas Winter [EMAIL PROTECTED] wrote:

Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING

2007-04-12 Thread Moises Silva
no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/ Kind Regards Moises Silva

Re: [asterisk-users] Asterisk-Java website

2007-04-12 Thread Moises Silva
Hum, I know Stefan, he is an asterisk-java dev, but he is not online right now, I will let him know ASAP. Thanks! On 4/12/07, Doug Garstang [EMAIL PROTECTED] wrote: Does anyone know who maintains the Asterisk-java web site at asterisk-java.org? The site seems to have been unavailable for a

Re: [asterisk-users] Trixbox 2.6.1.13 OpenR2

2008-11-29 Thread Moises Silva
There are probably other OpenR2 users that can help you in asterisk-r2 mailing list (http://lists.digium.com/pipermail/asterisk-r2/) I have not tried Trixbox, but the first release of OpenR2 will be next week, probably that will help to have Trixbox a proper support. Moy On Sat, Nov 29, 2008 at

Re: [asterisk-users] async agi question

2008-12-05 Thread Moises Silva
Hello Henrik, I have not used Asterisk from a user perspective lately, but, when I added the async agi functionality, I used to control this using a manager redirect action to the same priority where the channel calls async agi, that will work like a break that re-enters the async agi loop .

Re: [asterisk-users] R2

2009-01-15 Thread Moises Silva
Is in the process of being merged. http://bugs.digium.com/view.php?id=12509 http://reviewboard.digium.com/r/40/ http://www.libopenr2.org/ Moisés Silva On Thu, Jan 15, 2009 at 9:44 AM, David fire ddf...@gmail.com wrote: hi i am reading about new codecs and new stuff to be added to asterisk.

Re: [asterisk-users] R2

2009-01-15 Thread Moises Silva
That's Digium's folks decision. It was said they wanted it for 1.6.3, but, that's not for sure, as I said, they will decide. On Thu, Jan 15, 2009 at 11:54 AM, David fire ddf...@gmail.com wrote: thanks for the answer. any idea in wich version it will be merged? thanks 2009/1/15 Moises Silva

Re: [asterisk-users] codec_dahdi and Asterisk 1.6.0.6

2009-02-27 Thread Moises Silva
1) How can I use codec_dahdi? Would it be useful when passing a call from one dahdi channel to another dahdi channel? It is used whenever you need G729 or G723 transcoding (or any other format supported by the Digium transcoding board). If you don't have a Digium transcoding board then you don't

Re: [asterisk-users] The Redirect hangups the call while playing a file

2009-03-30 Thread Moises Silva
Hello, Which Asterisk version are you using? I was unable to reproduce your problem with Asterisk 1.6.0.3, also please post details about your dial plan extensions. Moy On Mon, Mar 30, 2009 at 7:13 AM, Jose Arias cyr2...@gmail.com wrote: Hi, I'm bringing this discussion here from

Re: [asterisk-users] async agi question

2009-04-02 Thread Moises Silva
Async AGI was never released for Asterisk 1.4.X, so probably the patch you used has a bug or something, do you still have the patch around? Moy On Thu, Apr 2, 2009 at 5:44 AM, cyr2...@gmail.com wrote: Hi Henrik, I would like to do the same thing you are doing here. I want to implement an

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