[asterisk-users] Queues with different technologies for members, like Khomp Driver
Guys, I saw in the Asterisk documentation (queues.conf) that members can register with technologies such as SIP, Dahdi and Location. But I have a specific need for members to be registered as Khompchannel. Ex: member => Khomp/b0L1/9200 But reloading module app_queue.so when I run the command "queue show", the member registered as Khomp appears as invalid: Khomp/b0L1/9200 (Invalid) I also read that I can try to load the driver used by members (in this case the Khomp driver) before app_queue.so to the Queue application can recognize that as a valid driver for a member. But I tried to do this with the "preload => chan_khomp.so" in modules.conf, but even then the Queue application continues to recognize the member as "invalid". Has anyone tried to do something like this, using other technologies to members of queues? PS.: Khomp is a kind of Channel Bank for E1 connections here in Brazil. Thanks in advance. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) & Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call notification on IP Telephone
I will try this tomorrow, that I will be with the phones near me. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) & Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 14:01, Danny Nicholas wrote: > That depends on too many things to answer in a short reply, but if you do it > "the right way", yes. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Varella > Sent: Monday, March 05, 2012 10:59 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Call notification on IP Telephone > > I already had think about this, but do you know if on the destination phone > B the caller ID of whom is calling the phone A, will be shown on display ?? > > -- > > Daniel Varella de Oliveira > Consultor de T.I. > Cel.: +55(21)8615-6050 > > Digium Certified Asterisk Administrator - (dCAA) > > Novell Certified Linux Administrator (Novell CLA) & Novell Data Center > Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 > > Linux Professional Certified - LPI > > Information Technology Infrastructure Library - ITIL Certified > > Cisco Certified Network Associate - CCNA > > > > On Mon, Mar 5, 2012 at 13:50, Daniel Varella wrote: >> I already had think about this, but do you know if on the >> destination phone B the caller ID of whom is calling the phone A, will >> be shown on display. >> >> Thanks. >> >> -- >> >> Daniel Varella de Oliveira >> Consultor de T.I. >> Cel.: +55(21)8615-6050 >> >> Digium Certified Asterisk Administrator - (dCAA) >> >> Novell Certified Linux Administrator (Novell CLA) & Novell Data Center >> Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 >> >> Linux Professional Certified - LPI >> >> Information Technology Infrastructure Library - ITIL Certified >> >> Cisco Certified Network Associate - CCNA >> >> >> >> On Mon, Mar 5, 2012 at 13:17, Administrator TOOTAI > wrote: >>> Le 05/03/2012 15:54, Daniel Varella a écrit : >>> >>>> Hi everybody, >>>> >>>> I'm seeking information on how to report an IP phone on a call >>>> that is occurring on another IP phone. >>>> >>>> Example: >>>> >>>> While the A phone is ringing, Asterisk sends a >>>> notification to a phone B on the call that is going to A, but this >>>> notification is displayed on the B phone display and the user does >>>> not need to hit anything to view the information. >>>> >>>> I'm working with "Siemens optiPoint 410 Economy" and "Yealink >>>> T22P" phones. >>> >>> >>> Use BLF if your phones support it. >>> >>> -- >>> Daniel >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call notification on IP Telephone
I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display ?? -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) & Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:50, Daniel Varella wrote: > I already had think about this, but do you know if on the > destination phone B the caller ID of whom is calling the phone A, will > be shown on display. > > Thanks. > > -- > > Daniel Varella de Oliveira > Consultor de T.I. > Cel.: +55(21)8615-6050 > > Digium Certified Asterisk Administrator - (dCAA) > > Novell Certified Linux Administrator (Novell CLA) & > Novell Data Center Technical Specialist (Novell DCTS) > SUSE Linux Enterprise 11 > > Linux Professional Certified - LPI > > Information Technology Infrastructure Library - ITIL Certified > > Cisco Certified Network Associate - CCNA > > > > On Mon, Mar 5, 2012 at 13:17, Administrator TOOTAI wrote: >> Le 05/03/2012 15:54, Daniel Varella a écrit : >> >>> Hi everybody, >>> >>> I'm seeking information on how to report an IP phone >>> on a call that is occurring on another IP phone. >>> >>> Example: >>> >>> While the A phone is ringing, Asterisk sends a >>> notification to a phone B on the call that is going to A, but this >>> notification is displayed on the B phone display and the user does not >>> need to hit anything to view the information. >>> >>> I'm working with "Siemens optiPoint 410 Economy" and >>> "Yealink T22P" phones. >> >> >> Use BLF if your phones support it. >> >> -- >> Daniel >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call notification on IP Telephone
I already had think about this, but do you know if on the destination phone B the caller ID of whom is calling the phone A, will be shown on display. Thanks. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) & Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA On Mon, Mar 5, 2012 at 13:17, Administrator TOOTAI wrote: > Le 05/03/2012 15:54, Daniel Varella a écrit : > >> Hi everybody, >> >> I'm seeking information on how to report an IP phone >> on a call that is occurring on another IP phone. >> >> Example: >> >> While the A phone is ringing, Asterisk sends a >> notification to a phone B on the call that is going to A, but this >> notification is displayed on the B phone display and the user does not >> need to hit anything to view the information. >> >> I'm working with "Siemens optiPoint 410 Economy" and >> "Yealink T22P" phones. > > > Use BLF if your phones support it. > > -- > Daniel > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call notification on IP Telephone
Hi everybody, I'm seeking information on how to report an IP phone on a call that is occurring on another IP phone. Example: While the A phone is ringing, Asterisk sends a notification to a phone B on the call that is going to A, but this notification is displayed on the B phone display and the user does not need to hit anything to view the information. I'm working with "Siemens optiPoint 410 Economy" and "Yealink T22P" phones. I Have searched some information on the Web, but without many details and solutions. I did some tests with the "sipsak" tool, but I was just able to send a notification to the Yealink phone (for tel. Siemens did not work) which appeared as a text message (like SMS), forcing the user having to press a few keys to obtain the information. And this is not the goal, but that notifies the phone screen, without the user having to press any key. Anyone have any idea or tip? Thank you in advance. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Digium Certified Asterisk Administrator - (dCAA) Novell Certified Linux Administrator (Novell CLA) & Novell Data Center Technical Specialist (Novell DCTS) SUSE Linux Enterprise 11 Linux Professional Certified - LPI Information Technology Infrastructure Library - ITIL Certified Cisco Certified Network Associate - CCNA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spy just a range of extensions
Alejandro,
I am using here the ExtenSpy() function, and it works very well.
I just change my dialout context to:
...
...
exten => _XXX,n,Set(SPYGROUP=callcenter)
...
...
And made a change to the callcenter context of the agents:
[monitoramento_callcenter]
exten => 88,1,Authenticate(12345|a|5)
exten => 88,n,ExtenSpy(,g(callcenter))
exten => 88,n,Hangup
The trick is on the SPYGROUP definition.
Best regards.
--
Daniel Varella de Oliveira
Consultor de T.I.
Cel.: +55(21)8615-6050
Novell Certified Linux Administrator (Novell CLA) &
Novell Data Center Technical Specialist (Novell DCTS)
SUSE Linux Enterprise 11
ID 10113097
Linux Professional Certified - LPI
LPI000143643
Information Technology Infrastructure Library - ITIL Certified
EXIN - 944759
Cisco Certified Network Associate - CCNA
On Wed, Aug 17, 2011 at 15:35, Alejandro Cabrera Obed wrote:
> Dear, I have to let some agents from a call center to spy/coach just a range
> of extensions. They must not spy extensions from boss and some other
> "important" people from my company.
>
> I have in extensions_additional.conf:
>
> [app-chanspy]
> include => app-chanspy-custom
> exten => 555,1,Macro(user-callerid,)
> exten => 555,n,Answer
> exten => 555,n,Wait(1)
> exten => 555,n,ChanSpy()
> exten => 555,n,Hangup
>
> and in extensions_custom.conf:
>
> [from-internal-custom]
> exten => 555,1,Macro(user-callerid)
> exten => 555,2,Authenticate(1234)
> exten => 555,3,Read(SPYNUM,agent-newlocation)
> exten => 555,4,ChanSpy(SIP/${SPYNUM))
>
> How can I let agents to just spy extensions 9000-9500 and no more ???
>
> Special thanks,
>
> Alejandro
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use AMD "Answering Machine Detect" ?
Yes, it worked !
Now I'm adjusting the AMD parameters to fine tune the recognition.
Thank you guys !
--
Daniel Varella de Oliveira
Consultor de T.I.
Cel.: +55(21)8615-6050
Linux Professional Certified
LPI000143643
Information Technology Infrastructure Library - ITIL Certified
EXIN - 944759
On Wed, Jan 7, 2009 at 14:02, Benoit wrote:
> Daniel Varella a écrit :
>> Hi everybody,
>>
>>Happy New Year !
>>
>>I'm trying to detect if a call was answered by a machine (linke
>> voicemail systems) or a human.
>>I would like to use AMD (Answering Machine Detect) command, but
>> with my configuration it was not possible get there.
>>
>>Follow my dialplan:
>>
>> exten => _[789].,1,NoCDR
>> exten => _[789].,n,Dial(SIP/${ext...@111,60)
>> exten => _[789].,n,AMD
>> exten => _[789].,n,NoOp(AMD Status is: ${AMDSTATUS})
>> exten => _[789].,n,Hangup
>>
>>What is happening is when the call is answered by the other part,
>> Asterisk doesn't go to the next level (exten => _[789].,n,AMD). So AMD
>> can't verify the call.
>>
>> How can I do this ? Any idea ?
>>
> iirc AMD is for the other way around, ie detect when a call receive on
> your box was made
> by an automated machine.
>
> In your case, the extension stop at the Dial() command, except if nobody
> answer your call.
>
> However, maybe you can try it with the M(x) argument of Dial():
>
> *M(*/x/*)*: Executes the macro (x) upon connect of the call (i.e.
> when the called party answers). IMPORTANT - The CDR 'billsecs' field is
> set to zero if the callee answers the call, but hangs up whilst the
> macro is still running (if the callee answers and the macro finishes,
> 'billsecs' contains the correct value).
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use AMD "Answering Machine Detect" ?
Hi everybody,
Happy New Year !
I'm trying to detect if a call was answered by a machine (linke
voicemail systems) or a human.
I would like to use AMD (Answering Machine Detect) command, but
with my configuration it was not possible get there.
Follow my dialplan:
exten => _[789].,1,NoCDR
exten => _[789].,n,Dial(SIP/${ext...@111,60)
exten => _[789].,n,AMD
exten => _[789].,n,NoOp(AMD Status is: ${AMDSTATUS})
exten => _[789].,n,Hangup
What is happening is when the call is answered by the other part,
Asterisk doesn't go to the next level (exten => _[789].,n,AMD). So AMD
can't verify the call.
How can I do this ? Any idea ?
Thanks in advance.
--
Daniel Varella de Oliveira
Consultor de T.I.
Cel.: +55(21)8615-6050
Linux Professional Certified
LPI000143643
Information Technology Infrastructure Library - ITIL Certified
EXIN - 944759
___
-- 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] Need Recording Solution in Asterisk
Well, I think this is a little difficult to deploy with a single system. Maybe you will need to deploy some distributed solutions (Asterisk or whatever) to have quality assurance, and coordinated to a central management. You will also need to verify how many concurrent calls between extensions on each location and intra-locations to determine how many trunk records to install and space to record all of these information too. Almost sure all these calls will need to pass through this system to being recorded. Do you alredy have some details of what kind of record and services of management your customer is asking for ? -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Linux Professional Certified LPI000143643 Information Technology Infrastructure Library - ITIL Certified EXIN - 944759 On Sat, Nov 22, 2008 at 16:51, David Cook <[EMAIL PROTECTED]> wrote: >> One of our client Bank has 900 employees working in different locations. >> They need to record all internal and external calls. Can any body suggest > Call Recording Solution for this >> requirement. We need to know the Hardware / Bandwidth and all > requirements and costing. > > Few questions first > 1. Why are they being recorded (business need)? > 2. Does the value of the recording remain constant over time or diminish? > 3. What criteria will you be required to retrieve the recording with? > 4. Do you expect users to retrieve their own recordings or make requests of > a records management operations staff? > 5. Does everything need to be on-line or near-line/off-line and do you > require a data management and migration solution? > 6. Do you need to do word spotting and trend analysis on the content of > these recordings (target marketing and customer service analysis typically)? > > Recording the call is quite easy. Storing it for retrieval which is > acceptable to the business under their potentially diverse requirements is > the tough part to nail down. > > There are commercial products like Witness out there which do a good job of > this at a premium price. If the business drivers have low impact, you could > simply record in asterisk and archive the files with some creative scripting > and database work. > > You said this is a bank so I'm presuming they will have a formal risk > analysis methods in place which would guide you through qualifying the > requirements. Check out what the IT/CIO folks have to help you out in this > manner. > > -dbc. > > > ___ > -- 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 > ___ -- 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] Require Billing solution for Calling Cards retail...
Hello Kashif, Do you have something already working ? Here in Brazil I've worked on some projects using Asterisk to make some passive call-centers receive calls from their remote customers. Is it what your customer is looking for ? About calling cards, Is something like pre-paid cards ? Do they have some system working, even without IVR ? Regards. -- Daniel Varella de Oliveira Consultor de T.I. Cel.: +55(21)8615-6050 Linux Professional Certified LPI000143643 On Sat, Jul 5, 2008 at 9:00 AM, Kashif Naeem <[EMAIL PROTECTED]> wrote: > Hello All, > > One of our French client is dealing in Wholesale termination business. Now > they are going to start retail of Calling Cards. They need complete IVR and > billing solution for it. Any one who has already provided such solutions > please contact. > > Regards, > > > -- > Kashif Naeem > Business Development Manager > Hadi Telecom > www.haditelecom.com > > Cell: +92 (0)345 4226006 > Office: +92 (0)42 5692766 > > Email: [EMAIL PROTECTED] > MSN: [EMAIL PROTECTED] > Gmail: [EMAIL PROTECTED] > Skype: kashif.naeem > > 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center SoftPhone with Auto Answer
Hi everybody, I'm coming back from a long time off silence in this list. I think this thread is going out of the focus of central conversation. The discussion about what is wrong and what is right in the law, to fire somebody who is not working behind the company rules, is not to be here. Well, about the question if is there some way to force somebody to answer a phone (like auto-answer) with no decision, the answer is Yes, there is. I have a customer that is using this configuration. The agents of call-center log into the callcenter system using the AgentLogin command, and start to hear some announce or some background music (you can configure to not play anything) and when a call comes, agent receives a beep of 1 second, to know that incoming call is starting, and automaticaly receives the call. To release this call, the agent can press *. Read more info here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentLogin Best regards. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tels.: +55 (21)3139-4091 / r. 108 +55 (11)3588-0802 / r. 108 www.tecnologiaip.com.br On Friday 21 September 2007 10:25, Tim Panton wrote: > You can configure Mexuar's Corraleta web-based softphone to autoanswer. > The user won't be able to unconfigure this, except by quitting the phone > (or page or browser) > > (Full disclosure - I'm a director at Mexuar and wrote much of the > code ;-) ) > > Tim. > > On 17 Sep 2007, at 16:11, Joao Pereira wrote: > > But still, the user can choose not to answer the phone. > > I want to force the users to accept the calls. > > > > Regards > > Joao Pereira > > > > Thiago Maluf wrote: > >> Ola Joao, > >> tem um modo do Asterisk fazer isso sim. > >> Entre em contato no meu GTALK por esse e-mail e eu te dou mais > >> informações. > >> Abs! > >> > >> Hi List, > >> The asterisk have one way to do it. > >> just put one script to discovery if this user is online or offline. > >> case is offline play one music. if not, call the user. > >> understand? > >> thiago! > >> > >> 2007/8/6, Joao Pereira <[EMAIL PROTECTED]>: > >>> Hello > >>> I need a Softphone with auto answer where users can't turn it off. > >>> Does someone knows a softphone where users can't turn the auto > >>> answer off? > >>> Or is there any way Asterisk could force the clients to answer > >>> the phone? > >>> > >>> Thanks > >>> Regards > >>> Joao Pereira > >>> > >>> ___ > >>> --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 > > > > ___ > > > > 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 list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > 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 list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users pgpLtDCBbbfWw.pgp Description: PGP signature ___ 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quintum Tenor DX 3020 problem to register on Asterisk
Hi Steve, Yes, I've tried this option "proxy calls only" enabled and disabled. But nothing changed. The curious is: I have another SIP server with OpenSER and it's happening too. I think the problem is not on the SIP servers (Asterisk or OpenSER), maybe something broken on the Quintum SIP support. I could not find any advanced information about SIP with Quintum. :-( Another idea ? -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br On Wednesday 24 May 2006 05:23, Steve Totaro wrote: > Daniel Varella de Oliveira wrote: > > Hi, > > > > I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box > > with SIP. Asterisk always returns "Username/Password mismatch". > > I've tried all configurations that was on the Quintum's manual, but no > > success. > > I've tested the same username and password with a Linksys (PAP2-NA) with > > the same asterisk box, and it worked fine. Where is the problem ? > > > > Looking for this on Google, I couldn't find any example of SIP > > configuration on Tenor DX. > > > > Does anyone have any example or a Quintum device working with SIP on > > Asterisk ? > > > > PS.: I'm using the last firmware of this Quintum -> P104-12-00 > > > > Please any idea will be great. > > > > Thanks in advance. > > Under the advanced configuration do you have a check in the "proxy calls > only" box? > > Thanks, > Steve > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users pgpdux8aDnX4X.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum Tenor DX 3020 problem to register on Asterisk
Hi, I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with SIP. Asterisk always returns "Username/Password mismatch". I've tried all configurations that was on the Quintum's manual, but no success. I've tested the same username and password with a Linksys (PAP2-NA) with the same asterisk box, and it worked fine. Where is the problem ? Looking for this on Google, I couldn't find any example of SIP configuration on Tenor DX. Does anyone have any example or a Quintum device working with SIP on Asterisk ? PS.: I'm using the last firmware of this Quintum -> P104-12-00 Please any idea will be great. Thanks in advance. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 one way audio using oh323
Mik, Your asterisk server is another machine of your GK ? You can start verifying if the traffic between the machines (related to RTP packets) is ok. Do you have firewall ? -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br On Monday 31 October 2005 08:05, mik sib wrote: > Hi all, > > through oh323 i can register to my gatekeeper and make > and receive calls. > > My gatekeeper routes the incoming call as well as the > outgoing. > > The problem is simply that i can't ear nothing from my > SIP ipPhones. I can ear my voice from a normal > telephone in my SIP phone but no viceversa. > > How can i debug this situation ? I've no errors in the > log or at the asterisk startup. > How to understand what's happening ? > I've tryed different phones also. > any idea ? > thank you very much > Mik > > > Here's my oh323.conf > Configuration of OpenH323 channel driver > -- > Version: 0.7.3 > Listening on address: 10.0.0.253:1720 > Gatekeeper used: [EMAIL PROTECTED] > (Registered) > FastStart/H245Tunnelling/H245inSetup: ON/ON/ON > Supported formats in pref. order: ulaw<0> > Jitter buffer limits (min/max): 20-100 ms > TCP port range: 1 - 2 > UDP (RAS) port range: 1 - 2 > UDP (RTP) port range: 1 - 2 > IP Type-of-Service value: 0 > User input mode: rfc2833 > Max number of inbound H.323 calls: 100 > Max number of outbound H.323 calls: 100 > Max number of simultaneous H.323 calls: 100 > Max call rate (ingress direction): 1.00/30 > Default language: en > Default music class: default > Default context: voip-h323 > > doing a call with the ip phone to the outside world > through the gatekeeper > > [2]WrapperAPI::h323_make_call: Making call. > [2]WrapH323EndPoint::MakeCall: Making call to > 0258115040 > [4]WrapH323EndPoint::CreateConnection: Creating a > H323Connection [32066] > [2]WrapH323Connection::WrapH323Connection: Creation of > WrapH323Connection based on user data. > [2]WrapH323Connection::WrapH323Connection: Call is > outgoing. > [4]WrapH323Connection::WrapH323Connection: > WrapH323Connection created. > [3]WrapH323EndPoint::MakeCall: Call token is > ip$localhost/32066 > [3]WrapH323EndPoint::MakeCall: Call reference is 32066 > [2]WrapH323Connection::OnSendSignalSetup: Sending > SETUP message... > [3]WrapH323Connection::OnSendSignalSetup: Setting > display name 0432281316 Fabio Violino > [3]WrapH323Connection::OnSendSignalSetup: Setting > calling party number test419 > [2]WrapH323Connection::OnAlerting: Ringing phone for > "0258115040" ... > [3]WrapH323EndPoint::OpenAudioChannel: Direction => > RECODER, Buffer => 320 > [2]WrapH323EndPoint::OpenAudioChannel: Media format: > FrameSize 8, FrameTime 8, TimeUnits 8 > [2]WrapH323EndPoint::OpenAudioChannel: Codec info: > FrameRate 160 > [2]WrapH323EndPoint::OpenAudioChannel: Packet size: > 160 > [2]WrapH323EndPoint::OpenAudioChannel: Frames per > packet: 20 > [2]WrapH323EndPoint::OpenAudioChannel: LID Codec > G.711-uLaw-64k > [3]WrapH323EndPoint::OpenAudioChannel: The sound > channel with the application is asterisk-oh323(fd=45) > [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object > initialized. > [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object > initialized. > [4]PAsteriskSoundChannel::PAsteriskSoundChannel: > Object initialized. > [3]PAsteriskSoundChannel::Open: os_handle 45, > mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize > 160 > [3]WrapH323EndPoint::OpenAudioChannel: Opened sound > channel "Asterisk" for recording using 1x320 byte > buffers. > [3]WrapH323Connection::OnEstablished: > WrapH323Connection [ip$localhost/32066] established > (FastStartDisabled/noH245Tunneling) > [3]WrapH323EndPoint::OnConnectionEstablished: > Connection [ip$localhost/32066] established. > [3]WrapH323EndPoint::GetConnectionInfo: > [ip$localhost/32066] RTP Media: > 10.0.0.253:10004-0.0.0.0:0 > [3]WrapH323EndPoint::OpenAudioChannel: Direction => > PLAYER, Buffer => 320 > [2]WrapH323EndPoint::OpenAudioChannel: Media format: > FrameSize 8, FrameTime 8, TimeUnits 8 > [2]WrapH323EndPoint::OpenAudioChannel: Codec info: > FrameRate 160 > [2]WrapH323EndPoint::OpenAudioChannel: Packet size: > 160 > [2]WrapH323EndPoint::OpenAudioChannel: Frames per > packet: 20 > [2]WrapH323EndPoint::OpenAudioChannel: LID Codec > G.711-uLaw-64k > [3]WrapH323EndPoint::OpenAudioChannel: The sound > channel with the application is asterisk-oh323(fd=43) > [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object > initialized. &g
Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?
It costs here more or less R$600,00 (about US$264,55) Our friend, Dave Cotton post a message with a good price for outside of Brazil. US$295,00 is a good price, I think. I know that guy in Sao Paolo (the correct is São Paulo), that the site http://www.thehightechstore.com/plugcell.htmlannounced. His name is Douglas Prado and he is the owner of Contacto Telecom company. Contacto is the unique distributor of Plugcell in region of São Paulo. If you contact him, tell about me (He knows me as Daniel ex-Nooracom company in Rio de Janeiro). Maybe you can get a discount on your negotiation. hehehehe. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br On Friday 28 October 2005 12:22, Tomasz Chmielewski wrote: > Daniel Varella de Oliveira schrieb: > > Tomasz, > > > > I'm from Brazil, and we are using here a solution that is based on a > > box where we can connect a GSM cellphone and use this directly to a > phone or PBX extension. > > > I think that you can use some Digium's card (FXS or FXO) on your > > server, connect this GSM box there, and route your cellphone calls > through this box. > > > There are boxes with just one channel and others up to six channels. > > They have a lot compatibilities with the most common cellphones. > > looks interesting. > > do you know by chance how much such a single-cell box cost (more or less)? pgpYL3LSzOWLh.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "GSM cards" / "mobile phone cards" for Asterisk?
Tomasz, I'm from Brazil, and we are using here a solution that is based on a box where we can connect a GSM cellphone and use this directly to a phone or PBX extension. I think that you can use some Digium's card (FXS or FXO) on your server, connect this GSM box there, and route your cellphone calls through this box. There are boxes with just one channel and others up to six channels. They have a lot compatibilities with the most common cellphones. Take a look to this site: http://www.zenitetecnologia.com.br/english/index.jsp Zenite is one of the best companies that build this boxes here in Brazil. I hope that I could help you. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 www.tecnologiaip.com.br On Friday 28 October 2005 10:26, Tomasz Chmielewski wrote: > I was wondering if there is something like that on this Earth: > > Some of our users are "mobile users" - they are rarely in one place for > longer than 15 minutes. > They use mobile phones a lot. > > From our mobile operator we have an offer which allows us to call for > free between our mobile phones. > > So the idea is to put a SIM card inside the Asterisk box, equipped with > a special card, a card which would be a mobile phone really. > > This would allow all office users to reach our mobile users without the > need of buying additional phones for the office users. > Office users would call Asterisk over IAX, and asterisk would call > "mobile users" using a free GSM/mobile. > > Does anyone have an idea if such cards exist, and if so, if they work > with Asterisk? pgpW1VfgS5q3m.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I want to send oH323 calls to our Quintum D3000 which is connected to a PSTN
Hi Ade,
An example of oh323.conf is attached, and the lines in the extensions.conf
that make the choice os this oh.323 channel is:
[globals]
GK => OH323/IP of your GK
[local]
ignorepat => 0
exten => _0021NXXX,1,Dial(${GK}/${EXTEN:1})
exten => _0021NXXX,2,Congestion
--
[ ]'s
Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
www.tecnologiaip.com.br
On Monday 26 September 2005 13:33, Ade Agbero wrote:
> Hello,
>
>
>
> I want to send oH323 calls to our Quintum D3000.
>
>
>
> I have installed oH323 but I need a working sample oh323.conf and
> extensions.conf, so that I can route specific calls to the Quintum using
> H323.
>
>
>
> For example our Asterisk box IP=192.168.10.100 and Quintum
> IP=192.168.10.101.
>
>
>
> Can anyone assist with a sample Extensions.conf and oH323.conf.
>
>
>
> Thank you,
>
>
>
> Ade.
>
>
>
>
>
>
>
>
> -
> Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with
> voicemail
;
; Configuration file of OpenH323 channel driver
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=2
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; "rtp.conf"
;
udpStart=1
udpEnd=2
;
; Enable fast start (yes,no).
;
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=no
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=3
libTraceLevel=3
;libTraceFile=stdout
libTraceFile=/var/log/asterisk-h323.log
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is the zone
name.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; ,
; ,
; GKID:
; @
;
gatekeeper=IP of your gatekeeper
;gatekeeper=DISABLE
;
; Set the gatekeeper password. If used, it enables H.235 access to gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
gatekeeperTTL=600
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
; Q931- Q.931 Keypad Information Element
; STRING - H.245 string
; TONE- H.245 tone
; RFC2833 - RFC2833
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
context=voip-h323
;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
context=all-aliases
alias=asterisk
alias=99001701
alias=99001702
;
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
Re: [Asterisk-Users] dialplan defenition (closer)
Joao, I don't think that number 81 is part of the dialed digits. Maybe this is an ID of this or something like this. I think that asterisk is not recognizing the first 2 digits, and passing just the others maybe is something related about ignorepat (like a "don't ignore pattern ?"). Anybody with any idea ? But analysing the confs, you are close to the right way... -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Wednesday 10 August 2005 13:43, Joao Pereira wrote: > Ok, I m getting to the point, > This route: > exten => _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) > Isn't working because the dialed number isnt maching _74XXX > > I putted Asterisk in "capi debug" mode and when I dial 74118 he says: > > > gnugk*CLI> capi debug > CAPI Debugging Enabled > -- CONNECT_IND ID=001 #0x0004 LEN=0078 > Controller/PLCI/NCCI= 0x401 > CIPValue= 0x10 > CalledPartyNumber = <81>118 > CallingPartyNumber = <01 83>118 > CalledPartySubaddress = default > CallingPartySubaddress = default > BC = <80 90 a3> > LLC = default > HLC = <91 81> > AdditionalInfo >BChannelinformation= default >Keypadfacility = default >Useruserdata = default >Facilitydataarray = <1c 23 9f aa 06 80 01 00 82 01 00 8b > 01 00 a1 15 02 02>6<5d 02 01 00 80 0c>JOAO PEREIRA > > Aug 10 17:25:22 NOTICE[1086933696]: chan_capi.c:1932 capi_handle_msg: > CONNECT_IND ID=001 #0x0004 LEN=0078 > Controller/PLCI/NCCI= 0x401 > CIPValue= 0x10 > CalledPartyNumber = <81>118 > CallingPartyNumber = <01 83>118 > CalledPartySubaddress = default > CallingPartySubaddress = default > BC = <80 90 a3> > LLC = default > HLC = <91 81> > AdditionalInfo >BChannelinformation= default >Keypadfacility = default >Useruserdata = default >Facilitydataarray = <1c 23 9f aa 06 80 01 00 82 01 00 8b > 01 00 a1 15 02 02>6<5d 02 01 00 80 0c>JOAO PEREIRA > > Aug 10 17:25:22 WARNING[1113294272]: pbx.c:1877 ast_pbx_run: Channel > 'CAPI[contr1/118]/0' sent into invalid extension 's' in context > 'default', but no invalid handler > -- MANUFACTURER_IND ID=001 #0x0005 LEN=0034 > Controller/PLCI/NCCI= 0x401 > ManuID = 0x4944 > Class = 0x70f000a > Function= 0x4f4a8300 > ManuData= O PEREIRA<81 29 00 00 00 25 1c 23 9f > aa 06 80 01 00 82 01 00 8b 01 00 a1 15 02 02>6<5d 02 01 00 80 0c>JOAO > PEREIRA<00 00 00 00 00 00 00 00 00 00 00 00 00> > > Aug 10 17:25:22 ERROR[1086933696]: chan_capi.c:2137 capi_handle_msg: > Command.Subcommand = 0xff.0x82 > -- INFO_IND ID=001 #0x0006 LEN=0019 > Controller/PLCI/NCCI= 0x401 > InfoNumber = 0x70 > InfoElement = <81>118 > > -- INFO_IND ID=001 #0x0007 LEN=0018 > Controller/PLCI/NCCI= 0x401 > InfoNumber = 0x18 > InfoElement = > > -- INFO_IND ID=001 #0x0008 LEN=0015 > Controller/PLCI/NCCI= 0x401 > InfoNumber = 0x8005 > InfoElement = default > > -- ALERT_CONF ID=001 #0x0004 LEN=0014 > Controller/PLCI/NCCI= 0x401 > Info= 0x0 > > -- DISCONNECT_IND ID=001 #0x000a LEN=0014 > Controller/PLCI/NCCI= 0x401 > Reason = 0x3490 > > --- >--- I believe that someware 74118 is being transformed in 118... but > the number that apears in this debug is > CalledPartyNumber = <81>118 > > How do I get this call? > I already tried: > exten => _81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) > exten => 81118,1,Dial(SIP/[EMAIL PROTECTED],30,r) > > but it never worked > Any ideas? > Thanks > Joao > > > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users pgpDbmxNCd7OR.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
Thanks Juan for the information. Altus, about the gatekeeper... It acts like a "DNS" on the h.323 world. Defining the gatekeeper: 1. Component of an H.323 conferencing system that performs call address resolution, admission control, and subnet bandwidth management. 2. Telecommunications: H.323 entity on a LAN that provides address translation and control access to the LAN for H.323 terminals and gateways. The gatekeeper can provide other services to the H.323 terminals and gateways, such as bandwidth management and locating gateways. A gatekeeper maintains a registry of devices in the multimedia network. The devices register with the gatekeeper at startup and request admission to a call from the gatekeeper. What I am trying to make is to change traffic communication between the two protocols (SIP and H.323). And I heard that is possible using a h.323 component on the Asterisk. I'm using oh.323 component from Innaccessnetwork and until now I'm on the right way. When it will finish I will post on the list the experience. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 12:39, altus wrote: > What is the difference? > Is it like register and registrar ? > If I make asterisk like a server and clients connect to it,is it a > gatekway? > And if I call another gateway its a gatekeeper ? > > On Thu, 2005-08-04 at 11:14 -0400, Juan Salas wrote: > > >From wiki... > > > > (http://www.voip-info.org/tiki-index.php?page=Asterisk+oh323+channels) > > > > "The second option is valid only in the case where a gatekeeper is used. > > OH323 supports only one gatekeeper (or none, but not multiple > > gatekeepers). OH323 itself only acts as H.323 Gateway. " > > > > As I look, asterisk didn't act like gatekeeper. > > > > JS. > > > > >Yes, it worked here. > > > > > >part of oh323.conf example: > > > > > >. > > >. > > >. > > >;- > > >; Configure H.323 aliases, prefixes and > > >; related ASTERISK's contexts > > >;- > > >[register] > > >; > > >; Aliases/prefixes associated with the default context > > >; defined in section [general]. > > >; > > >;alias=asterisk > > >;alias=123 > > >; > > >; Aliases/prefixes routed in "all-aliases" context. > > >; > > >context=all-aliases > > >alias=asterisk > > >alias=99001701 > > >alias=99001702 > > >. > > >. > > >. > > > > > > This defines h.323 id and the aliases for each channel. > > > > > > So, now I would like to know if asterisk can support h.323 gateway > > >registration, like SIP. Can a h.323 gateway register on asterisk ? > > >Thanks > > > > > >-- > > > > > >[ ]'s > > > > > >Daniel Varella de Oliveira > > >Tecnologia IP Ltda > > >Tel.: +55 (21)2495-0936 / r. 108 > > >www.tecnologiaip.com.br > > > > > >On Thursday 04 August 2005 10:54, Juan Salas wrote: > > >> Yes you can. > > >> Try with oh323 module: > > >> > > >> http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.h > > >>tml > > >> > > >> With this module you can register your asterisk with a gatekeeper. > > >> > > >> Regards. > > >> > > >> JSalas. > > >> > > >>> -Mensaje original- > > >>> De: altus [mailto:[EMAIL PROTECTED] > > >>> Enviado el: Thursday, August 04, 2005 5:30 AM > > >>> Para: asterisk > > >>> Asunto: [Asterisk-Users] h323 > > >>> > > >>> > > >>> Good day all > > >>> Can I register asterisk as a h323 client,like in sip where you have > > >>> register => > > >>> > > >>>-Mensaje original- > > >>>De: Daniel Varella de Oliveira [mailto:[EMAIL PROTECTED] > > >>>Enviado el: Thursday, August 04, 2005 10:42 AM > > >>>Para: Asterisk Users Mailing List - Non-Commercial Discussion > > >>>Asunto: Re: [Asterisk-Users] h323 > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users pgpWdXLGBp6VW.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
Yes, it worked here. part of oh323.conf example: . . . ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in "all-aliases" context. ; context=all-aliases alias=asterisk alias=99001701 alias=99001702 . . . This defines h.323 id and the aliases for each channel. So, now I would like to know if asterisk can support h.323 gateway registration, like SIP. Can a h.323 gateway register on asterisk ? Thanks -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 10:54, Juan Salas wrote: > Yes you can. > Try with oh323 module: > > http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html > > With this module you can register your asterisk with a gatekeeper. > > Regards. > > JSalas. > > > -Mensaje original- > De: altus [mailto:[EMAIL PROTECTED] > Enviado el: Thursday, August 04, 2005 5:30 AM > Para: asterisk > Asunto: [Asterisk-Users] h323 > > > Good day all > Can I register asterisk as a h323 client,like in sip where you have > register => pgpj3jix6qCO3.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
