Re: [asterisk-users] SIP provider and NAT

AST -- FW NAT -- CARRIER sip.conf externip=PUBLIC IP GW OF ASTERISK nat=route localnet=IP AND LOCAL MASK(ex .192.168.0.0/255.255.0.0) ;Carrier example [Carrier] type=friend host=CARRIER IP fromdomain= CARRIER IP context=incoming disallow=all allow=g729 canreinvite=no insecure=very Good Luck

[asterisk-users] MixMonitor and Queues

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1

[asterisk-users] RV: MixMonitor and Queues

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1

[asterisk-users] MixMonitor Problem

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1

[asterisk-users] Realtime MOH

Hi, I'm having troubles using music on hold with realtime (ast 1.6.0.1). Everything seems ok, but no clases are show. If I try to make a cal nothing is played and says theres no moh class. Musiconhold.conf [general] cachertclasses=yes ; use 1 instance of moh class for all users

[asterisk-users] MOH Realtime Problem

Hi, I'm having 2 problems: 1) MOH in realtime is not working, I have configured it but never go to look at the database, no warning or error found and I can do a query using realtime and the family from the cli. 2) I have SIP phones via realtime, if I register one of them and a

Re: [asterisk-users] MOH Realtime Problem

My second problem is resolved, qualify=yes did the trick. I'm still having problems with MOH De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sebastian Enviado el: Friday, November 21, 2008 9:09 PM Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] MOH Realtime

Re: [asterisk-users] MOH Realtime Problem

Someone?? Any idea?? De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sebastian Enviado el: Friday, November 21, 2008 9:09 PM Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] MOH Realtime Problem Hi, I'm having 2 problems: 1) MOH in realtime

Re: [asterisk-users] Disabling Call-Waiting

You can try call-limit = 1 in sip.conf for each phone. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Elliot Murdock Sent: martes, 25 de noviembre de 2008 11:04 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Disabling Call-Waiting Hello! I have a few

Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

I think call-limit is for 1 outbound and 1 inbound so if you get a call you wont get other but if you do an outbound call an incoming call will be allowed. Maybe you can configure 1 line in your phone or try Check device state before make the call in extensions. -Original Message- From:

Re: [asterisk-users] Disabling Call-Waiting

You can try: exten = 1234567,1,Dial(SIP/515) exten = 1234567,101,Dial(SIP/516) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mikhail (Plus Plus) Sent: martes, 25 de noviembre de 2008 02:08 p.m. To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] MOH Realtime

Anybody was able to set it up?? I can't make it work, any idea?? Ast 1.6.0.1 Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Disabling Call-Waiting

If is deprecated how do you treat a queue (realtime), that has to have just one call for agent?? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: martes, 25 de noviembre de 2008 03:37 p.m. To: Asterisk Users Mailing List -

Re: [asterisk-users] Disabling Call-Waiting

looking at the DEV_STATE function (available separately on Asterisk-1.4). It will tell you the status of the phone before you call the Dial() application. __Yehavi: 2008/11/25 Sebastian [EMAIL PROTECTED] If is deprecated how do you treat a queue (realtime), that has

[asterisk-users] CDR Hangupcause

Hi, I'm trying to get HANGUPCAUSE on my cdr the problem I'm facing is that this option: endbeforehexten=yes is not working at least on asterisk 1.6.0.1, so if I put yes o no I cant set CDR value with that value. It seems to finish the CDR record before h is executed. I'm using

[asterisk-users] MOH Realtime

Is working on 1.6.0.1?? someone was able to make it work? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call

Have you snniffed the packages? It seems to be some kind of difrerence on the notify, try to sniff a packet ok and then one with error Enviado desde mi iPhone El 28/11/2008, a las 01:01 a.m., OCG Technical Support [EMAIL PROTECTED] escribió: I’m trying to get my Windows Mobile 6 phone

Re: [asterisk-users] MOH Realtime

Sorry the spelling but im writing on the phone with spanish corrector :) Enviado desde mi iPhone El 28/11/2008, a las 02:29 a.m., dinhtrung [EMAIL PROTECTED] escribió: Hi Sebastian, http://bugs.digium.com/view.php?id=11196 Nguyễn Đình Trung --- QiS Technologies, ltd. Tel: 0168 528 7522

[asterisk-users] Parking calls

Hi, How can I park a call from dialplan and get going?? Example: 1. Answer 2. While follow = false 3. ParkCall 4. Checksomthing à follow = true 5. Endwhile 6. UnParkCall 7. Go on….. The idea is let the call waiting while I do

Re: [asterisk-users] Parking calls

Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Sent: lunes, 01 de diciembre de 2008 11:58 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Parking calls Hi, How can I park

Re: [asterisk-users] Parking calls

: Tuesday, December 02, 2008 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking calls It is not a parking solution. Sebastian wrote: Any idea? Please I need advice. Thanks! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf

Re: [asterisk-users] Parking calls

. Any idea?? Thanks! Sebastian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: miércoles, 03 de diciembre de 2008 03:04 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking calls

Re: [asterisk-users] Parking calls

-users] Parking calls On Wed, Dec 3, 2008 at 7:27 PM, Sebastian [EMAIL PROTECTED] wrote: The thing is I have to wait checking a database value to change the state, that duration is not long, but on any case I don't know when will be ready to go on. If I use MusicOnHold app the dialplan get stuck

Re: [asterisk-users] Parking calls

, Dec 4, 2008 at 1:25 AM, Sebastian [EMAIL PROTECTED] wrote: I don't understand how can I solve my situation with this Ok, a simplified sample (i used PHP because i use it daily, but any language is good): context incoming { _X. = { Answer(); System(channel-waiting.php ${CHANNEL

[asterisk-users] MOH Realtime

Someone could make it work??? I tried everything and there's no way I can make it work! Someone can help me? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] top posting again [was: Re: CDR Design]

I agree, you end up searching through the mail to read what you want. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barton Fisher Sent: viernes, 05 de diciembre de 2008 11:31 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] top

[asterisk-users] Func_ODBC question

: Error -1 in FETCH [UPDATE Tabla set campo = 4356] Any idea why is this?? The query works fine, I just wanto to know if the warning can cause any problem to me. Thanks!! Sebastian ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] a problem on Ubuntu with Asterisk

Have you do just asterisk before try to reconect to cli?? Try asterisk - to see if is crashing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez Sent: miércoles, 10 de diciembre de 2008 01:12 p.m. To: [EMAIL PROTECTED]; Asterisk

Re: [asterisk-users] Country numbering plan resources

You forgot Uruguay I can give you the info if you want :) Enviado desde mi iPhone El 13/12/2008, a las 01:10 a.m., Michael mich...@networkstuff.co.nz escribió: Is there any good free / accurate online resources with detailed country numbering plans? Failing that let's get something

[asterisk-users] Queue Question

, Sebastian ___ -- 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] Queue Question

- Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Question Sebastian wrote: Hi, In queues realtime, when the queue start and when it ends. I mean, for example to calculate service level, how many calls, etc. If I want to start the queue from with 0 calls, etc, how

Re: [asterisk-users] Queue Question

. Thanks Regards, Sebastian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian Sent: lunes, 15 de diciembre de 2008 09:00 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re

[asterisk-users] question about ringinuse

Just a silly question that I'm not sure. Ringinuse is working with IAX in 1.6??? like in sip?? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] Compile problems

The fax error seems to be problem of spandsp version. What version are you using??? Enviado desde mi iPhone El 07/03/2009, a las 09:26 p.m., Remco Barendse aster...@barendse.to escribió: Hi all I don't know what went wrong but i no longer seem to be able to compile asterisk. I first do

Re: [asterisk-users] Compile problems

, Sebastian wrote: The fax error seems to be problem of spandsp version. What version are you using??? I use the latest IAXMODEM 1.2.0, the changelog of it says update spandsp to 20080725 snapshot However, i never asked Asterisk to compile with fax support, can i disable fax support somewhere

Re: [asterisk-users] 1.4.23 + Realtime Queues/Agents NOT via SIP

IAX2 also support InUse, is a good choice for Agents at home because IAX is nat friendly :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peter Beckman Sent: martes, 10 de marzo de 2009 12:24 p.m. To:

[asterisk-users] SetVar (CDR var) from cli

Hi, I'm using 1.6.0.5 I'm trying to set CDR(userfield) for example from the cli, I look at the channel variables and I can see the new status, but que it hang-ups the CDR doesn't have this value. I'm using mysql backend for cdr Any idea? Thnks

Re: [asterisk-users] SetVar (CDR var) from cli

is not working. Any idea?? Regards Sebastian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: jueves, 12 de marzo de 2009 05:57 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] Queue Realtime agents LOGIN for ami

Is there any AMI action that logs a realtime agent? I mean, if you send it, queue_log and queue_member get the corresponding inserts. Regards Sebastian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

[asterisk-users] MOH Realtime

I've finally got it working on 1.6.0.6, but it seems to be a problem: Situation: Queue realtime musiconhold class = prueba I have default class on musiconhold.conf When a call is made to the Queue checks that is not on memory so goes to the db, I had 2 situations: 1) If digit

[asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Forget about this. Is still working. From: Sebastian [mailto:s...@adinet.com.uy] Sent: viernes, 13 de marzo de 2009 10:05 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: TRANSFER EVENT ON QUEUE_LOG Hi, Anyone knows if TRANSFER event on queue_log

Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG

Sent: viernes, 13 de marzo de 2009 10:14 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG Sebastian wrote: Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer

[asterisk-users] 1.6.2 beta 1 crash

Hi, I'm starting testing 1.6.2 beta. CentOs 5.2 I found my first crash, first I have [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201]:

Re: [asterisk-users] 1.6.2 beta 1 crash

Seems to be a crash with func_odbc. Let me know what info you need to check it. Thnks From: Sebastian [mailto:s...@adinet.com.uy] Sent: viernes, 20 de marzo de 2009 10:37 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: 1.6.2 beta 1 crash Hi, I'm

Re: [asterisk-users] 1.6.2 beta 1 crash

Discussion Subject: Re: [asterisk-users] 1.6.2 beta 1 crash On Friday 20 March 2009 20:36:38 Sebastian wrote: [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201

Re: [asterisk-users] 1.6.2 beta 1 crash

Issue: 0014716 With just DON't OPTIMIZE Works realtime but func_odbc doesn't crash. I put the message of realtime not working with debug malloc on the bug. Let me know if you need more info. Regards Sebastian -Original Message- From: asterisk-users-boun...@lists.digium.com

[asterisk-users] H323plus homepage down?

Anybody knows why is down? Or if has been moved to another page?? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] H323plus homepage down?

-users] H323plus homepage down? Sebastian wrote: Anybody knows why is down? Or if has been moved to another page?? ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] make script 1.6.0.6 breaks up, need help!

Are you installing on a 64bit OS?? Which Os are you using?? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: domingo, 22 de marzo de 2009 05:59 p.m. To: Asterisk Users Mailing List -

[asterisk-users] AMI IAXPeers

Is there any reason why IAXPeers output is different from SIPPeers output? The response has no Eventlist: start Ej. Response: Success Eventlist: start Message: Peer status list will follow Event: PeerEntry Channeltype: SIP ObjectName: 1001 ChanObjectType: peer IPaddress:

Re: [asterisk-users] Asterisk 1.6.2 Beta

Use: console dial Regards, -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian Sent: viernes, 24 de abril de 2009 01:07 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.6.2

[asterisk-users] Outgoing Queues

Anyone thought about something like outgoing queues? I mean, having same info that has for inbound queues but for outbound calls, and grouping members there. For example, before using dial application put an app outqueue that get all the statics. Talked time, member status, last call, completed

Re: [asterisk-users] Outgoing Queues

Ok, and that’s exactly what I mean monitoring outbound groups, so you can have realtime info for monitoring. And as with queues have the ability to reset the statics for monitoring porpouses. -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] callcenter / dialer / predictive dialer / vicidial program is now open

ONGs -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: lunes, 18 de mayo de 2009 06:12 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] callcenter /

Re: [asterisk-users] Asterisk manager login with java not working

I would recommend you to use Asterisk-Java library has support for manager, agi, etc. http://asterisk-java.org/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: lunes, 08 de junio de

Re: [asterisk-users] Setting CDR(userfield) from Macro called from feature doesn't work with cdr_mysql

Check this issue, seems related https://issues.asterisk.org/view.php?id=14662 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown Sent: martes, 30 de junio de 2009 11:33 a.m. To:

[Asterisk-Users] RxFax Asterisk possible bug?

this problem (with Asterisk 1.0.9). So I don't believe this is an error caused by myself. -- With kind regards, Sebastian Berm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Operator Console(s)/Shared Call Appearances

, with one agent. Furthermore, if your company starts to grow, and more receptionists that have to answer the phone are needed, it's quite easy, all you have to do is add a sip account, one agent and add that agent to the existing queue. (About 2 minutes...) -- Sebastian Berm iPronto Communications

Re: [asterisk-users] 2 Problems with 1.6.2

Have you solved this issue? When I restart the machines I can't make an outgoing DAHDI call until I get an incoming call on that same line. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: viernes,

Re: [asterisk-users] Continue dialplan is source channel hangs up

F([[context^]exten^]priority): When the caller hangs up, transfer the called party to the specified destination and continue execution at that location. Also just F will continue to the next priority on the dialplan. De: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Which voicemail file format is the most widely understood ?

and Linux at least. Don't know about Mac though. I used to use gsm - but only some apps play it on Linux - and I could only find QuickTime Player to play it in Windows. Simple 'wav' appears to be incompatible (at least not directly) with Windows. Sebastian Suggestions ? Regards

Re: [asterisk-users] Changing voicemail.conf file format list

in voicemail.conf, it will work equally well. However, your users might not be very pleased :-) Sebastian As this statement is written, it seems there is a trap too complex to detail which surprises me a bit. Comments ? Regards

Re: [asterisk-users] Problems with audio

of the Grandstream phones we were using solved the problem. I don't know if this is your case as well though. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Digest Username/auth name mismatch ‏

, a hardphone? It looks like the client is sending the full a...@192.168.0.1 instead of just as the username. Sebastian Register From: sip:a...@192.168.0.1;tag=644056924 To: sip:a...@192.168.0.1 Call-ID: 2457796...@192.168.0.2 CSeq: 125 REGISTER Contact:sip:a...@192.168.0.2:5060

Re: [asterisk-users] DTMF

phone). Also, it looks like 'inband' only works with ulaw and alaw. http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Sebastian Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Call Recording Questions

it? The space is not incompatible with either Samba or Linux filesystem. However, is the ~ character part of the filename you are creating? If it is, that is definitely an illegal/reserved character in the Linux file systems. Sebastian exten = s,n(donothing),MacroExit Thanks Dan

Re: [asterisk-users] one way audio for xlite clients behind NAT

On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Re: [asterisk-users] Purpose of qualify=yes

it doesn't matter if they are unavailable for a period of time. At least this is the conclusions I have reached. So it seems that sometimes qualify is not a good idea. Sebastian Chris Owen wrote: We have a tenant who has been having issues with a congested connection and in trouble shooting

Re: [asterisk-users] one way audio for xlite clients behind NAT

your X-lite configuration - and maybe try to ulaw or alaw as the only codec at both ends? Sebastian On Thu, Sep 16, 2010 at 1:36 PM, Sebastian s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue

Re: [asterisk-users] 3rd party app store

-source/not-for-profit and commercial operations or intentions. You get a not-very-pleasant smell about it when the two start to intermingle to the point where you can't tell where one ends and the other begins. Sebastian

Re: [asterisk-users] Digest Username/auth name mismatch ‏

On 09/21/2010 04:26 AM, t. k wrote: Hi Thanks for help. I will try to help. But others might know more. What SIP client are you using - a softphone, a hardphone? It looks like the client is sending the full at 192.168.0.1 instead of just as the username. Sebastian That's

Re: [asterisk-users] Asterisk as a distributed paging system

nodes working at the same time. I only used one. I did a write-up here - see if you find any useful information: http://forum.voxilla.com/asterisk-support-forum/asterisk-public-announcement-system-loud-ringer-bell-49339.html Sebastian I need the system to be resilient to any network partition

Re: [asterisk-users] looking for a better ATA

or usb fxo interface is not what you are looking for - or doesn't fit your requirements? Sebastian Thanks for any input Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] How to connect asterisk PBX to PSTN

. Sebastian Some vendors are there for these PSTN cards like Digium, Sangoma, Openvox. Good luck:) On Thu, Oct 14, 2010 at 7:05 PM, Jigar Joshi jiga...@gmail.com mailto:jiga...@gmail.com wrote: Hello community, I have successfully set up asterisk free PBX server and I am also

[asterisk-users] IAX2 call dropped when a second call comes in

be much appreciated. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] Mobile Phones and Asterisk

using ver. 1.6. Thanks in advance. I'm not sure I understand your setup. Are you using SIP for trunking, or for extensions? Are you calling a normal mobile phone, or a SIP client on a mobile phone? Sebastian regards, RYAN ICASIANO From: asterisk

Re: [asterisk-users] Mobile Phones and Asterisk

xxx in front of ${extension}? You shouldn't need them. Just pass ${extension} - which is the number you dialled on the phone. Sebastian I hear a busy tone, after the 10 sec. timeout it returns NOANSWER, as defined in my DIAL func. I also tried getting the DEVICE_STATE exten =s,3,NoOp(SIP

Re: [asterisk-users] Mobile Phones and Asterisk

in Asterisk. Sebastian I achieved this successfully by emulating it via a softphone, when I call a softphone and it is currently engaged in a call, asterisk returns BUSY in DIALSTATUS and will automatically fallback to the next step in the dialplan. But this is not the case when applying

Re: [asterisk-users] Mobile Phones and Asterisk

is not available. To Asterisk, that message is the same as somebody answering the line. Same in France and Spain - as far as I've seen. Sebastian - still no answer that pots line is hung up and call drops back into the original extension's vm. (I have not run into a problem with answer detection, only

Re: [asterisk-users] Asterisk 1.8 Multiple Parking Lots

like it below)? Sebastian /exten = 2011,3,ParkedCall(2011)/ // /exten = 2012,hint,park:2...@parked/ /exten = 2012,1,Wait(1)/ /exten = 2012,2,Set(CHANNEL(parkinglot)=parkinglot_A/ /exten = 2012,3,ParkedCall(2012)/ // /exten = 2021,hint,park:2...@parked/ /exten = 2021,1,Wait(1)/ /exten

[asterisk-users] Random call drops on IAX2

appreciated. I'm not sure where else to look to get more relevant information. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Random call drops on IAX2

anybody? On 11/10/2010 06:51 PM, Sebastian wrote: Hello list, I have an Asterisk setup with the following details: 1. 3 x internal extensions / sip hardphones - Grandstream 2000 2. 2 x internal extensions / dahdi cordless phone 3. 1 x 2 FSX ports OpenVOX pci card 4. 1 x internal sip

Re: [asterisk-users] Asterisk Playback sound dropping on linphone

linphonec as well - and haven't found another console sip phone either. I'd be interested if there is another one. Sebastian Thank you, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] IAX2 and INVAL packets

) when I tried. So not easy to replicate. Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] IAX2 and INVAL packets

On 11/18/2010 10:38 PM, Tilghman Lesher wrote: On Thursday 18 November 2010 14:01:49 Sebastian wrote: Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been

Re: [asterisk-users] IAX2 and INVAL packets

On 11/18/2010 08:01 PM, Sebastian wrote: Is anybody here familiar with the meaning of INVAL packets for IAX2? Every few days I get a dropped outgoing call in the middle of the conversation (the outgoing call has been connected for few minutes) when an incoming call comes in. The log reads

Re: [asterisk-users] Moving asterisk from one network to another.

, they might contain the IP's of external and internal networks in the config files. Your firewall (if you have one) might contain IP's and network masks. It depends on how the box was originally setup. Sebastian It is assumed that DB is on the same box. Asterisk box has got Asterisk running

Re: [asterisk-users] Log and forward calls to cellphone?

diversion setup. Hope the above helps, Sebastian Thanks guys. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

to look at it. Sebastian Looking forward to your analysis. Regards, Bruce On Tue, Dec 28, 2010 at 3:58 PM, Moises Silva moises.si...@gmail.com mailto:moises.si...@gmail.com wrote: On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: I

Re: [asterisk-users] load balance with 2 wan connections

about load balancing and failover setups if I remember correctly. Just a thought, Sebastian On 12/25/2010 11:01 PM, Dave George wrote: Server will have two fix public ips. Dave Original Message Subject: Re: [asterisk-users] load balance with 2 wan connections From

Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?

/messages or /var/log/syslog contain device not accepting address or USB device is disconnected error messages. Sebastian At least that is the case with mine. If you don't have it then it's I guess very specific to this atom board. I know someone else added something along the lines irq

Re: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)

is larger, you should also add maybe the calling extension to the file name - so that you don't have two files with the same name - if two extensions try to call the same external number at exactly the same time (seems unlikely to me). Sebastian Any advise? Regards Bilal

Re: [asterisk-users] Log and forward calls to cellphone?

- but this is not compulsory). [...] As a last alternative - a slight improvement on the above. If you can get a smartphone with Android - which would let you run SIP over 3G - you should have true free voice divert. Thanks Sebastian for the tip. The goal is to 1) have clients call the usual

Re: [asterisk-users] Log and forward calls to cellphone?

On 01/04/2011 01:55 PM, A J Stiles wrote: On Tuesday 04 Jan 2011, Gilles wrote: Thanks Sebastian for the tip. The goal is to 1) have clients call the usual landline number instead of asking them to try a cellphone in case no one's home, 2) get Asterisk to handle the call, 3) have

Re: [asterisk-users] Log and forward calls to cellphone?

find the above detail very useful. Many thanks, Sebastian [...] 2. what smartphone supports installing an SIP + OpenVPN clients? Without OpenVPN lots off, IPhone, Android, Nokia, Windows mobile, ... Best SIP client integrated with mobile are Nokias (E series for instance). I'm running HTC Hero

Re: [asterisk-users] Log and forward calls to cellphone?

Hi, On 01/05/2011 10:49 AM, Administrator TOOTAI wrote: Le 04/01/2011 20:50, Sebastian a écrit : Hi, On 01/04/2011 03:24 PM, Administrator TOOTAI wrote: Le 04/01/2011 11:50, Gilles a écrit : [...] It looks like getting a 3G smartphone with SIP + OpenVPN + unlimited Internet plan would

Re: [asterisk-users] OpenVPN + SIP configuration?

, and Linphone, Twinkle and Ekiga on Linux as clients - if it helps. Sebastian Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] Busy Detection on Analog Lines

Hi, I'm having an issue with busy detection, the busy is not being detected. Asterisk: 1.6.2.13 DAHDI: 2.4.0 Chandahdi: busydetect=yes, busycount=2 Indications zone = us, with the modifications for my country for busy: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)

Re: [asterisk-users] Variable. AMI and dialplan

this may be related with: https://issues.asterisk.org/view.php?id=14662 El 28/03/2011 10:20, Sherwood McGowan escribió: Don't know then, that's all I've got far ya today mate, sorry On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote: I did use Action: Getvar when i read it back in AMI.

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