Re: [asterisk-users] Asterisk and fax

Asterisk+fax Regards, Ricardo Carvalho. On Feb 13, 2008 5:49 PM, voip crazy <[EMAIL PROTECTED]> wrote: > I want to receibe the fax via mail and send faxes via web interface and a > digital send and receibe fax list. > > Voipcrazy > > 2008/2/13, Giorgio Incantalupo <[EMAI

Re: [asterisk-users] SNMP monitoring

I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Regards, Ricardo Carvalho. On Thu, Feb 14, 2008 at 5:30 PM, Adrian Marsh <[EMAIL PROTECTED]> wrote: > Hi All, > > > &

Re: [asterisk-users] SNMP monitoring

Maybe you'r right and newt isn't really necessary. I just read somewhere that those dependencies were needed, I've installed them and it worked... Try to only install the other ones and if res_snmp gets compiled without it, great! Regards, Ricardo Carvalho. On Fri, Feb 15, 2

[asterisk-users] logging the estimated RTT using SIP

Is it possible in Asterisk 1.4 to log by somehow the estimated roundtrip time (RTT) between server and some peer, which Asterisk computes based on the sending of OPTIONS and the receiving of the responses to those OPTIONS? Regards, Ricardo Carvalho

[asterisk-users] two lines written in CDR for each failed call in asterisk 1.4

this behavior expected in Asterisk 1.4 or didn't I port correctly my dialplan syntax from 1.2 to 1.4? Since I want CDR to be written like it was in 1.2, can this "feature" be disabled somehow? Regards, Ricardo Carvalho. ___ -- Bandwidth a

[asterisk-users] T.38 passthrough in asterisk 1.4

ic of those FAXes go out using the same IP of my Asterisk server through the SIP trunk I have established with the telco I have subscribed? This is because, this telco only accepts all traffic (SIP+RTP) sent by the same IP. Regards, Ricardo Car

Re: [asterisk-users] T.38 SIP Issues

I made some tests with FAX in Asterisk 1.4 using T.38 between two ATAs connected to legacy FAX machines, and realized that only SIP can make passthrough in the server while RTP go direct between endpoints. Is it possible for RTP data stream also to make passthrough in Asterisk? Thanks, Ricardo

Re: [asterisk-users] Realtime replication!!!!!

Take it: http://www.onlamp.com/pub/a/onlamp/2006/04/20/advanced-mysql-replication.html?page=1 Regards, Ricardo Carvalho. On Wed, Mar 26, 2008 at 10:45 AM, Al Baker <[EMAIL PROTECTED]> wrote: > Could you point a link to the DUAL MASTER Replication. > I swear i have been all over

Re: [asterisk-users] Calling users to the external domain using Asterisk

n => _.,9,HangUp() exten => _.,10,Goto(noturi-default,${EXTEN},1) exten => h,1,HangUp() [noturi-default] ;(your dialplan) Regards, Ricardo Carvalho. On Thu, Mar 27, 2008 at 7:47 AM, Aadilkhan Maniyar <[EMAIL PROTECTED]> wrote: > Hi All, > > > > I am a newbie to

Re: [asterisk-users] Calling users to the external domain usingAsterisk

can also trigger some Set(CDR(userfield)=SRV call from ${SIPCHANINFO(recvip)}) so that in your mysql CDR table be written which calls got sent by IP to any SIP URI. Regards, Ricardo Carvalho. On Fri, Mar 28, 2008 at 12:00 PM, Aadilkhan Maniyar <[EMAIL PROTECTED]> wrote: > Thanks for

[asterisk-users] T.38 VoIP providers

By your experience, please someone tell me which T.38 capable VoIP SIP providers have you tested with success sending and receiving FAX with Asterisk 1.4. Thanks, Ricardo Carvalho. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Authenticate() application and CDR

ome kind of BUG? (I'm using Asterisk version 1.2.17) Regards, ------ Ricardo Carvalho ITEC / IRICUP / Reitoria UP tel: +351220408108 (Ext: 5219) e-mail/sip: rjcarvalho[at]reit.up.pt -- __

Re: [asterisk-users] Authenticate() application and CDR

Thanks Atis, You've helped a lot. Regards, Ricardo. -- Ricardo Carvalho ITEC / IRICUP / Reitoria UP tel: +351220408108 (Ext: 5219) e-mail/sip: rjcarvalho[at]reit.up.pt -- Atis Lezdins wrote: > O

Re: [asterisk-users] Grandstream GXP2020 / 2000

rects this issue. Older ones hadn't so much good hands-free speaker, but recent ones have a better DSP from Texas Instruments. Althow they're not the best choice in the market (like Cisco or Polycom), they represent a good price/quality ratio. Regards, R

[asterisk-users] faster timeout in ENUMLOOKUP() function

comes faster? Or any ideas to avoid this problem? Regards, Ricardo Carvalho. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing li

Re: [asterisk-users] Realtime extensions configuration - calling user filtering

As much I as can tell, Asterisk version 1.2 doesn't support the "ex-girlfriend logic" that you ask. I didn't test that feature with 1.4 releases, maybe they already implement it. Regards, Ricardo Carvalho.. On Nov 20, 2007 2:51 PM, Tomasz Zieleniewski <[EMAIL PROTECTE

Re: [asterisk-users] Realtime extensions configuration - calling user filtering

". In the same script you can even do some "asterisk -r -x "extensions reload"" command, and then you'll have your own realtime extensions working with the "ex-girlfriend logic" you wanted! I implemented this way because I had the same problem as you... :) Regar

Re: [asterisk-users] Softphone to be installed on the Mobile

Here's one sip softphone for mobiles you can give a try: http://www.minisip.org/ Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] Get IP address of an incoming or outgoing SIP call

You may take a look at the SIPCHANINFO(recvip) function. With it, you can even start logging into CDR the IPs of incoming and outgoing calls. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk

Re: [asterisk-users] Asterisk behind a PIX firewall?

messages got rewritten by its public IPs, it should have, or else you'll never get it working right. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or u

Re: [asterisk-users] Finding the status of an extension

those monitored extensions. Regards, Ricardo Carvalho. ___ --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] hostname in MySQL CDR records

, Ricardo Carvalho. ___ --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] Shared line appearance phones?

I don't know if I understood you right, but can't that be solved with call queues? http://www.voip-info.org/wiki/index.php?page=Asterisk+call+queues http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Regards, Ricard

Re: [asterisk-users] Do While loop

You can try something like this: exten => _X.,1,SET(condition=${RAND(1,2)}) exten => _X.,2,GotoIf($[${condition} = '1']?1:3) exten => _X.,3,SET(Result is 2) Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provide

[asterisk-users] Logging in and off sessions in the dialplan

icate', but as far as I know, with this application the user meeds to dial his pin at each call he whats to make, and that not what I need! Some ideas? Thanks, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com--

[asterisk-users] Cisco power injector with GXP2000 phones

;t it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visi

Re: [asterisk-users] Logging in and off sessions in the dialplan

Thanks Gordon, I'll give it a try with astDB. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/ma

Re: [asterisk-users] Cisco power injector with GXP2000 phones

is kind of detail. If someone does, please tell. Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailma

[asterisk-users] how can qualify=yes trigger some external event?

Hi all, The option qualify=yes allows Asterisk to check if it can reach the peer. If the device does not answer within the time-out period, Asterisk considers the device off-line for future calls. Is it possible to use this feature to trigger some external event, in case of failed reply from t

[asterisk-users] ENUMLOOKUP well succeeded but callee server unreached

I use ENUM lookup in my dialplan before sending calls through my PSTN trunk. One problem arises... When ENUMLOOKUP finds an SIP contact for that e164 number, Asterisk dials that contact, but when the remote server that should answer the call is down, or the IP link is down for some reason, the d

[asterisk-users] Forward to my phones the domain of the CALLERID in incoming URI calls

Is there a way I can forward to my phones the domain of the CALLERID in the CALLERID(number) field of INVITE messages, when some call arrives to my Asterisk? What happens in my architecture is this: INVITE [EMAIL PROTECTED]

Re: [asterisk-users] Forcing Dial application to skip if called server is unreachable

ov > Evariste Systems > Web: http://www.evaristesys.com/ > Tel: +1-678-954-0670 > Direct : +1-678-954-0671 > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCR

[asterisk-users] sharing phone registration information between asterisk servers

Is it possible to share SIP phones registration information between two different asterisk servers, that share the same realtime MySQL DB? Regards, Ricardo. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users maili

[asterisk-users] how to force asterisk to read registration information from DB

B table to know if destination phone is registered? Regards, Ricardo Carvalho. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/m

[asterisk-users] how to load phone registration information

Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk "thinks" those phones are already registered? This would be very usefull for a redundant server... Regards, Ricardo Carvalho. ___ --Ban

Re: [asterisk-users] how to load phone registration information

emory and isn't read from DB! Is there any way that I can force Asterisk to read sip_buddies realtime DB table to know if destination phone is registered? Regards, Ricardo Carvalho. On 7/12/07, Ricardo Carvalho <[EMAIL PROTECTED]> wrote: Is it possible to load phone reg

[asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension

Dear all, How may I configure my extensions.conf so that only the boss's secretary can call the boss through his extension, all others when dial his extension only makes the boss's secretary phone ring, not his. If she wants, she can transfer the incoming call to the boss dialling his extensi

[asterisk-users] Distinct call permissions for each user

Dear all, How may I configure my extensions.conf to stablish different PSTN access permissions for each user, letting for example user_A make only local calls and user_B make local and long-distance calls? I guess it can be done using include of other contexts, but how exactly? someone please

Re: [asterisk-users] Distinct call permissions for each user

Thanks Luki, that's exactly what I was looking for, I'll give it a try... Regards, Ricardo. Luki wrote: someone please give me one example? [locals] exten => _NXX,1,Macro(outcall,${EXTEN}) [longdistance] exten => _1NXXNXX,1,Macro(outcall,${EXTEN}) [macro-outcall] exten => s,1,Dial

[asterisk-users] Asterisk with Radius users authentication

Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch, an still open branch of Digium Issue Tracker "SIP peer authentication on an externa

Re: [asterisk-users] Asterisk with Radius users authentication

? Thanks once again, Ricardo. yusuf wrote: Ricardo Carvalho wrote: Dear all, I've searched the web about Asterisk with Radius integration for user authentication, and got a bit confused... I see that there have been some work around it, there is PortaOne's Radius client patch

[asterisk-users] Monitoring which users are online in realtime

Hi all, Is there a way to keep track in Asterisk of which phones are online in realtime using some MySQL DB table for exemple, much like "sip show peers" does in the CLI? Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] Ex-Girlfriend syntax and RealTime Extensions

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any oth

[asterisk-users] multiple phones registered for the same user

Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the "sip show peers" in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone call

Re: [asterisk-users] multiple phones registered for the same user

om: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Wednesday, February 28, 2007 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] multiple phones registered for the same user Dear all, I've noticed that when

Re: [asterisk-users] multiple phones registered for the same user

Too bad... Thanks for all replays. Regards, Ricardo. Eric "ManxPower" Wieling wrote: Ricardo Carvalho wrote: Can't I register multiple phones with the same user/password? That's what I pretend to do, not ring groups... No, you cannot register multiple phones with th

Re: [asterisk-users] FAX using T38

Only Asterisk 1.4.0 has chan_sip.c with T.38 code in it. Although it still doesn't work because it's full of bugs. Seems to me that developers just pasted the T.38 patch code from the branch developing that issue, and nothing else have done to improve it. It has to be debugged. Regards, Ricard

Re: [asterisk-users] Zaptel 1.4.0

Try this: /etc/init.d/zaptel start Than do lsmod |grep zaptel and it should show zaptel loaded Ricardo. Mike Hammett wrote: I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make install and I don’t see any errors. This is out of my modprobe.conf: install tor2 /sbin/mod

[asterisk-users] BLF not working with Asterisk 1.4.0

Dear all, I've implemented BLF for use with some Grandstream GXP-2000 phones and it works fine in 1.2.x versions of Asterisk, although I tested it with version 1.4.0 and it doesn't work! Has the needed syntax changed for configure BLF for this version of Asterisk? It it a bug of this version?

[asterisk-users] sip.conf outboundproxy

#x27;t the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Am I doing something wrong or is this the real behaviour of the outboundproxy variable in sip.conf? Best Regards, Ricardo Carvalho. ___ -- Bandwidth and Colocation P

Re: [asterisk-users] sip.conf outboundproxy

users] sip.conf outboundproxy > > On 24 Mar 2009, at 17:51, Ricardo Carvalho wrote: > > > Hi, > > > > I'm trying to enable sip.conf outboundproxy support in version > > 1.4.20.1 of Asterisk, but for the tests I made, every calls, even > > internal SIP calls

Re: [asterisk-users] sip.conf outboundproxy

So, does anyone ever used outboundproxy in sip.conf with success? Does it only send OUTBOUND calls via the proxy and not also internal extension calls via that proxy? Best Regards, Ricardo. ___ -- Bandwidth and Colocation Provided by http://www.api-digi

Re: [asterisk-users] sip.conf outboundproxy

Thanks Kevin. Although it doesn't fit my needs, thanks for the explanation. I guess I'll really have to combine Asterisk with OpenSer to do what I want. Ricardo. On Thu, Mar 26, 2009 at 1:07 PM, Kevin P. Fleming wrote: > Ricardo Carvalho wrote: > > > Does it only s

[asterisk-users] How to match wild card inside a GoToIf?

How can I match wildcards inside a GoToIf? I have something like this, but it doesn't work: [default] exten => _2,1,Macro(outcall,${EXTEN}) [macro-outcall] exten => s,1,GotoIf($["${ARG1}" = "220408XXX"]?2:3) exten => s,2,Hangup Any ideas? Regards, Ricardo. ___

Re: [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension

-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Friday, January 26, 2007 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Only secretary can call the boss, all others only re

[asterisk-users] unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting

Dear all, In my Asterisk 1.2.17 architecture different levels of permissions are established using different contexts that hierarchically include more permissive contexts until default context is reached. In default context there are only local extensions, only in more restricted contexts ther

[asterisk-users] Asterisk Pickup with more than one argument

Dear all, I tried to use the following sintax to implement call pickup in Asterisk 1.2.17 with no success: exten => _**5219/5215,1,Pickup(5219) exten => _**5219/5215,2,Pickup(220408108) exten => _**5219/5215,3,Hangup Asterisk seems to just do the first priority command (Pickup(5219)) and if

[asterisk-users] Call Pickup with more than one argument

Dear all, Does Pickup application accept multiple extensions pickup syntax, like the following line? Pickup(extension1&extension2&...) I've tried it in Asterisk 1.2.17 but it doesn't work. Does it work in Asterisk 1.4 already? Or is any other way in any version of Asterisk that I can use to

[asterisk-users] SIP OPTIONS triggering some action in case of no reply

Hi, Is it possible to implement some kind of alarmist triggering some action, by sending SIP OPTIONS messages regularly to check that other peer is still online? I'm using Asterisk version 1.2.11 which I know it doesn't have any SNMP module, just 1.4 branch is being developing one; but is it a

[asterisk-users] SER + Asterisk PSTN calls don't hung up

Hi, I'm deploying a SER + Asterisk architecture, where SER is used to manage acc, users database and sip routing, and Asterisk is used for voicemail and PSTN gateway. The system is already able to make and receive calls from the PSTN, although, only after the call has been established it can b

Re: [asterisk-users] SER + Asterisk PSTN calls don't hung up

and also: - exten => _0.,2,Busy exten => _0.,3,Hangup - Ricardo. Ricardo Carvalho wrote: Hi, I'm deploying a SER + Asterisk architecture, where SER is used to manage acc, users dat

[asterisk-users] Asterisk Real Time and sip.conf file used at the same time

Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] Portuguese sound files available?

Hi, I've been searching for sound files in Portuguese language to use in Asterisk for example for voicemail, but I couldn't find anything... Does anyone know where I could find them for download, if there is such thing already? Regards, Ricardo. _

[asterisk-users] Asterisk t38passthrough

Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax machine. Every time I send a fax, it gets sent using codec G711, and never T.38. I added the following parameters in the [general] section as well as in device configurations: t3

Re: [asterisk-users] Asterisk t38passthrough

gh Perhaps a stupid suggestion... but did you make sure that the ATA had the T38 selected in the GUI? bp On 8/24/06, *Ricardo Carvalho* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstrea

[asterisk-users] Does anyone use T.38?

Does anyone use T.38 for fax? If you use it, what hardware / software do you use? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.co

Re: [asterisk-users] Asterisk t38passthrough

rsion 1.2.4 works for me. If anyone deployed with success those versions that I didn't make to work, please tell me! Regards, Ricardo. Ricardo Carvalho wrote: Hi, I've installed Asterisk t38passthrough branch and I'm using one Grandstream ATA to connect Asterisk to a Fax m

Re: [asterisk-users] Handytone 286 T.38 SDP parameters

Dear Jason, Only version 1.2.4 of the Asterisk-t38 branch worked for me. Matybe your problem could be that. Try to install version 1.2.4, it should work. Regards, Ricardo. Jason Kim wrote: Hello, I installed trunk version of asterisk. I'm testing T.38 fax. My configuration is FaxMachine

Re: [asterisk-users] Handytone 286 T.38 SDP parameters

38pt_tcp=no Regards, Ricardo. Ricardo Carvalho wrote: Dear Jason, Only version 1.2.4 of the Asterisk-t38 branch worked for me. Matybe your problem could be that. Try to install version 1.2.4, it should work. Regards, Ricardo. Jason Kim wrote: Hello, I installed trunk version of aste

[Fwd: Re: [asterisk-users] Asterisk t38passthrough]

You can download the patch for t.38 passthrough from the URL: http://bugs.digium.com/file_download.php?file_id=9335&type=bug Regards, Ricardo. Patrick wrote: On Tue, 2006-08-29 at 12:50 +0100, Ricardo Carvalho wrote: Finally it's working! I was doing everything well, the pr

Re: [asterisk-users] asterisk t.38 fax failed

No, T.38 doesn't work with Asterisk. Only works with Asterisk t38passthrough patch that you can find at URL: http://bugs.digium.com/file_download.php?file_id=9335&type=bug For me it only worked well with patch for version 1.2.4 of Asterisk. Regards, Ricardo. Kokfoo Soo wrote: Is T.38 fax

Re: [asterisk-users] asterisk t.38 fax failed

= t38UDPRedundancy T38FaxMaxDatagram = 400 udptlfecentries = 3 udptlfecspan = 3 Good luck, Ricardo. Kokfoo Soo wrote: Ricardo, Thanks, could you please share some of your t.38 passthrough configuration in sip.conf and also udptl.conf? Thanks, */Ricardo Carvalho <[EMAIL PROTECTED]>/*

[asterisk-users] distinguishing users by their domain

In extensions.conf I want to implement a dial plan that distinguishes the users that wish to dial a PSTN number by their own domain, so that [EMAIL PROTECTED] goes out to PSTN by a different DID than [EMAIL PROTECTED] I tried the following line, but that doesn't distinguish between domains, an

Re: [asterisk-users] distinguishing users by their domain

So... does anybody know how can I do this? Maybe using a way to distinguish users not by their username, but by other fields of SIP INVITE messages? Regards, Ricardo. Ricardo Carvalho wrote: In extensions.conf I want to implement a dial plan that distinguishes the users that wish to

[asterisk-users] Can Asterisk bind on multiple ports?

Can Asterisk bind on multiple ports? I wish I could in my sip.conf make Asterisk bind different ports per different context, so that calls coming in udp port 5060 would fall in one context and calls coming in port 5061 fall in other different context. Is that possible? How can I edit my sip.con

[asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right

I have tested "Grandstream Budgetone 102" and "Grandstream Budgetone 200" and with both, if they are called from a caller that is an alphanumeric user, their display shows a unintelligible name impossible to figure out who is calling!! If the caller is a numeric one, in both phones their displa

Re: [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right

Thanks Tom, That's too bad... right now that I was thinking about buying them to a mass deployment environment... Regards, Ricardo. Tom Vile wrote: They only do numeric callerid. On 9/11/06, *Ricardo Carvalho* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wr

Re: [asterisk-users] Grandstream Budgetone phones don't show

I guess this functionality will be in the future added to new firmware releases don't you people think so? Ricardo. Doug Lytle wrote: These phones aren't capable of alphanumeric entries, only numeric. Doug Tom Vile wrote: They only do numeric callerid. __

Re: [asterisk-users] Grandstream Budgetone phones don't show

EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote: I guess this functionality will be in the future added to new firmware releases don't you

Re: [asterisk-users] Grandstream Budgetone phones don't show alphanumeric caller right

uy wrote: The lcd in the current budgetone series cannot support alphnumeric display. Craig - Original Message - From: "Ricardo Carvalho" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, September 12, 2006 8:1

[Asterisk-Users] Problem with VoicemailMain

Hi, I'm running SER with Asterisk, and I've configured VoicemailMain like this: exten => 201,1,VoicemailMain(@default) exten => 201,2,Hangup() Although, after any user enter his voicemailmain mailbox, when the phone is hung up, the call still continues running in Asterisk, because I can see i

[Asterisk-Users] Asterisk keeps running after hungup untill I press #

Hi, I'm running SER with Asterisk, and I've configured VoicemailMain like this: exten => 201,1,VoicemailMain(@default) exten => 201,2,Hangup() Although, after any user enter his voicemailmain mailbox, when the phone is hung up, the call still continues running in Asterisk, because I can see i

[asterisk-users] Thomson 2030

Hi all, Does Thomson 2030 hardphone has the feature of supporting more than one user registered at the same time? I heard not... But I think that's weird because it has 4 profiles... Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easyne

Re: [asterisk-users] Re: Can you explain why multiple registration is an important (missing) feature ?

You can use Asterisk along with Ser. Asterisk for advanced features like Voicemail and gateway, and Ser for routing SIP messages, Registrar, acc, etc. Take a look at: http://www.voip-info.org/wiki-Asterisk+at+large It works!! Regards, Ricardo. Tomislav Parčina wrote: In article <[EMAIL PRO

[asterisk-users] Voicemail adjustments

Hi all, Some questions about Asterisk Voicemail adjustments I want to make: - how can I limit the number of voicemail messages stored per user in their voicemail folder? (to expire voicemail after a specified number of days I know that there is in /contrib/scripts one script to do that) - ho

[asterisk-users] Does Asterisk 1.4 going to support realtime ex-girlfriend logic?

Hi all, I was deploying Realtime Extensions when I realised that Realtime Asterisk yet doesn't support ex-girlfriend logic, what made me abandon that implementation! Does Asterisk 1.4 going to support that feature? Regards, Ricardo. ___ --Bandwidth

[asterisk-users] voicemail usernames can't begin with "j" letter?

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file f

Re: [asterisk-users] SIP_HEADER function; what names are available?

Any news on this thread? I also need to know the way to get the R-URI from sip INVITE messages received by Asterisk, through ${SIP_HEADER()}. Thanks in advance, Ricardo. kjcsb wrote: I have read the wiki about the SIP_HEADER function (http://www.voip- info.org/wiki/index.php?page=Asterisk

Re: [asterisk-users] voicemail usernames can't begin with "j" letter?

ce Reeves wrote: What version of * are you running? I have several "j" usernames in voicemail.conf under SVN-branch-1.2-r37458M. On 10/20/06, *Ricardo Carvalho* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Dear all, I've configured Asterisk Voic

Re: [asterisk-users] voicemail usernames can't begin with "j" letter?

Thanks for all that replayed, the problem is solved! Regards, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-us

Re: [asterisk-users] anti ex-girlfriend

Has far as I know, Asterisk doesn't support ex-girlfriend logic in realtime extensions so far. Regards, Ricardo. Pezhman Lali wrote: Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received

Re: [asterisk-users] Glitches in sound every time that Asteriskreceives reINVITEs

In fact as far as I know, Asterisk stands in the middle of calls, breaking one transaction and initiating another to the other side, doing the bridge between them... Although good in some cases like permitting to start a new transaction to the next hop changing codecs, in my case I don't need t

[asterisk-users] special characters in alphanumeric extensions

Hi all, I use alphanumeric names as extensions in my Asterisk architecture, which are the username part of the e-mail of each person at my site. Because Asterisk was primarily built to use numeric extensions, I'm having some problems with people that have usernames with dots between letters,

[asterisk-users] problem with redirects

Dear all, My architecture is having some problems with redirects. In the following diagram is shown a simple erroneous test. When someone dials from the PSTN, signalling of the incoming call is passed to Asterisk which routes to SIP Express Route (Ser), and then Ser routes to the phone. The us

Re: [asterisk-users] problem with redirects

OK, to simplify the reading I'll resume my problem... Is there a way to make Asterisk send a call to Ser witch reroutes it back to the same asterisk server ,without resulting in a "loop detected" error in Asterisk? Thanks, Ricardo. ___ --Bandwid

[asterisk-users] FAX using T38

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success

[asterisk-users] How to disable the 482 Loop Detected messages sent by Asterisk

Is there a way to make Asterisk don't send "482 Loop Detected" error messages and continue with the transaction that is taking place? Thanks, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRI

[asterisk-users] Call Transfers in SER + Asterisk architecture

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfer

Re: [asterisk-users] Call Transfers in SER + Asterisk architecture

would be easier for me to try to help you on this. Does asterisk is registred into SER , or you have trust based relationship between them? On 11/23/06, *Ricardo Carvalho* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Hi, I'm deploying a SER + Asterisk

[asterisk-users] any enum test number of e164.arpa tree?

Can anybody please tell me any ENUM test DID from e164.arpa tree, which I can use to test some features? Thanks, Ricardo. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a li

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