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Juan Manuel Acevedo

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Everything posted by Juan Manuel Acevedo

  1. Juan Manuel Acevedo


    Hi: There are a lot of free and commercial applications with features as IVR, predecitive dialing, etc. Most of them are homologated for asterisk using Activa for Asterisk, and another ones can work with "Generic TAPI". Does the PBxnSIP is one "Generic TAPI", so I can use it to work with PBXnSIP as the IP-PBX for such applications? Thanks. Juan Acevedo acevedo1@une.net.co
  2. Henry: Can you tell me how Can I get SIPfire? Thanks Juan Acevedo
  3. Hi: I had the same problem and I have found that the best way to do inter-domain calls is: 1. Define all dial planas as "global" 3. Insert in such dial plans one entry, as the first of the table, with the option "Try Loopback" and * as Pattern and Replacement (both) 4. Set OFF the option "Loopback detection" as a global set. All calls between domains will occur. Juan Acevedo consultorit@umi.com.co
  4. Hi: The callas between domains is working by now, The problem I had was I have not assigned the Dial plan to the extension is calling, but: The calls extension-to-extension are working fine, I can reach the AA of another domain but when i dial for ask some extension, the call can not be completed, also occurs if I call to one Agent gruop from another domain. Another point is that the *87 for call piockup does not work in the domain extensions I have set "Try Loopback" at dial plan and I hev set "NO" the loopback detection at global settings. Any body can help me? Thanks Juan Acevedo Consultorit@umi.com.co
  5. Hi: I ahve followed the instructions at wiki: <LI>The mode "Try Loopback" is useful when you have several domains on the same PBX and you want to send a call from one domain to another without the need of an external SIP proxy. In this mode, the PBX will first check if there is a match of the pattern, and then calculate the destination using the replacement pattern. If the destination matches a telephone number (starting with "+") on the local system, then the PBX will send the call back to the system. Note: You must turn "Loopback detection" off (see Overall System Settings) and you must have at least one UDP socket (not bound to any specific address or bound to the loopback address or [::1]). Also, you need to have the "country code" (Domain->Settings page) set on both the domains. I have one domain named dominio1 with extension 120 and AA 100, and another domain named dominio2 with 200 as AA and extension 220. I can make calls between accounts of each domain I have configured at dominio1 one dial plan with Try Loopback and 2* as pattern and 2* as replacement. I have set off loopback detection off at overall System Settings and I and i have created in dominio1 one trunk outbound with outbound proxy When I call to 200 from one extension of the dominio1 I can see tha INVITE sip:200@dominio1, it means that the dial plan and trunk is not working. What is wrong? Thanks Juan Acevedo consultorit@umi.com.co
  6. Hi: I fact I have the same requirement, I need to accept a call from a predictive dialer so PBXnSip uses his own PSTN trunk to make the call. In previuos verision (2) it was possible. Thanks Juan Acevedo consultorit@umi.com.co
  7. Jim: You must to create just one trunk, and set in each AA as alias, the DID you want drive to, works also for extensions, hunt groups and agent group Juan Acevedo Consultorit@umi.com.co
  8. hi: I need to handle several domains and make calls between them, but i don't find the instructions at WIKI, can some body tell me when they are? Thanks Juan Acevedo
  9. Hi: I have two ethernet cards in my server, one for the local private network with the Ip address, and another one for establish a link with one ITSP with the address, my domain is localhost. I have set one outbound proxy trunk to make the link with the ITSP because it does not ask for registration. I need send the from to ITSP as from sip:mynumber@, I can do it changing my local host by but my internal calls are lost. How can I modify the from sip:mynumber@localhost to sip:mynumber@ WITHOUT change my domain name? Thanks Juan Acevedo consultorit@umi.com.co
  10. HI: I need to contact some reseller in Barcelona. Thanks Juan Manuel Acevedo My email: consultorit@umi.com.co
  11. Hi: A few days ago 3CX has announced his Gateway to Skype so that calls made to Skype users can be drawed to 3CX extension , does PBXnSIP have some plans about it? Thanks and best regards Juan Acevedo consultorit@umi.com.co
  12. Thanks; can you tell me where can I to find information about?, I dont see at WIKI Juan Acevedo
  13. Thanks, Yes I can pickup the call dialing *87+extension number, but my specific situation is that not allways whom needs to pickup the call knows what extension is ringing. I have made one Agent group and I can pickup the call dialing *87+Agent Group number, but it only works when the calls incomes via the Agent Group, not if the caller dial the specific extension number For example: I have 3 extensions: 501, 502 and 503 I have the Agent group 500 with 502 and 503 as agents. One call incomes to number 500, rings at 502 and 503, from 501 I dial *87500 and the call is picked up But if the income call goes to 502 directly or trhu one AA, and when the call is ringing, from 501 I dial *87500 and the answer is "service unavailable" Other IPPBX, permits configure at extension one "Call Pickup Group" so I can pickup any call is ringing in any extension of the grupo without diference of the way the call incomes. Thanks
  14. Hi: I need to pickup a call ringing in one extension of one of my colleagues in my same departament. There is not any grup, hunt or agent, but we are working for the same departament, for example accounting. When I dial *87 I can pickup any call is ringging in whole company but I need to pickup only the calls are ringing at the extensions of my departament´s colleagues. It is possible to do that?, If yes How? Thanks and best regards Juan Acevedo consultorit@umi.com.co
  15. Hi: Thanks for the information, "Send call to extension" sends all calls to the extension. Aliases works sendong the call by checking the called number. What I need is to route the call depending on the CALLER ID. Thanks
  16. Hi: I have one sip registration to my Telephony carrier, and it sends to me the caller ID of the user is calling me, it is possible to route the call to one extension depending on the caller ID I receive? I don´t see the way to do that, when I have used a gateway it has the way to configure a table based on the caller ID, but I don't see nothing similar at trunk I configure to register my PBXnSIP with my carrier. Thanks and Best Regards Juan Acevedo consultorit@umi.com.co
  17. Matt: I have installed more than 500 Quintum gateways, a lot of them as gateways for PBXnSIP and all of them works fine!, the best fetaure is it have one DSP for each voice channel. www.quintum.com Best regards Juan Acevedo consultorit@umi.com.co
  18. Hi: Does any body knows if exist some predictive dialer software that can be a SIP User Agent? In this way it can get register from PBXnSIP and use the PBXnSIP trunks to make calls and when the call be connected it can transfer it to some agnt group. Would be nice! Thanks Juan Acevedo acevedo1@une.net.co
  19. Hi Fred: I have the same problem, change at agent group setting, the Agent Selection algorithm from Random (default) to Ring longest idle first or to the 3th option, it works. Best Regards Juan Acevedo acevedo1@une.net.co
  20. hi: Waht is the max numbers of entries in one dial plan? I need define up 300, will be this possible? Thanks Juan Acevedo acevedo1@umi.com.co
  21. Hi: Is It possible to use PCI cards with PBxnSIP?, as same asterisk does, if anybody have done please tell us Best Regards Juan Acevedo acevedo1@une.net.co
  22. Hi: It think the problem is not sending mials, PBXnSip sends ok the email with the attachment when is one voice mail, but when it needs to send one mail with the attachment with the recording just it does not. I cann see it with one sniffer when the recording ends, just don´t send it. Thanks
  23. HI: I have installed Version 3.01 beta when can I to find how configure: In order to solve this problem without having to buy an expensive certificate, we added the option to turn TLS off. The name of the setting is "smtp_starttls". Thanks Juan Acevedo
  24. In fact I see at recordings folder the file, but is not sended to the mail box Juan Acevedo
  25. Hi: I am trying of to record a current conversation using the record button of one Snom 320; I have configured the email address at extension of the Snom 320, and I m using a license with call recording, also I have used a version without call recording. I am expecting a mail to such email address attached with the wav produced by the recording, but it does not happen, all the other messages like voice mails arrive to the email address, but file with recording does not . I renember some previous version where this feature was working fine, by now I use Any body can tell me what is happening? Thanks and best regards. Juan Acevedo consultorit@umi.com.co acevedo1@une.net.co
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