Re: [asterisk-users] Integrate Astreisk with SIP interface
what SIP interface means? could you make an example On May 12, 2013, at 4:04 AM, luke devon luke_de...@yahoo.com wrote: Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -- _ -- 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] Integrate Astreisk with SIP interface
what you mean by interface? if you want connect sip phone with asterisk there are 2 file to modify 1. sip.conf 2. extensions.conf. for creating sip user add following in sip.conf [ivr_user] defaultuser=ivruser ;username for sip phone secret=ivruser ;password for sip phone context=ivrcontext for more read examples in sip.conf for ivr add following in extensions.conf [ivruser] exten 123,1,Playback(your ivr file goes here) exten 123,n.Hangup for more read examples in extensions.conf reload asterisk register sip phone with asterisk dial 123 from sip phone hope this will help you. On Sun, May 12, 2013 at 4:04 AM, luke devon luke_de...@yahoo.com wrote: Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -- _ -- 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] time zone setting in asterisk
you can try to set usegmtime=no in cdr.conf On Sun, May 12, 2013 at 3:40 AM, Joseph syscon...@gmail.com wrote: Which file in Asterisk have a setting for time zone? When asterisk record incoming call in Master.csv the time is 6hr. ahead. I'm on: Canada/Mountain zone -- Joseph -- __**__**_ -- 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-**usershttp://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] ISP trunk session ID [SOLVED]
Hello I solved it and I leave the solution here for those who need it because this modems software is retarded, it does not allow to put the VOIP (VSPA_SIP) client to any other interface so I couldn't do the trick to connect it back to my own asterisk server on LAN and monitor what packets it send out. It only connects through the ONT (optical) link through it's wan adapter, and that cannot be bridged again. So basically I can't even use it as an ATA later for my own pbx... So I needed to register at the PBX in this format: register = fromuser@fromdomain:secret:authuser@host:port/extension instead of: register = authuser:secret@host:port/extension Best Regards,Sergej 11.05.2013, 10:30, "Sergej Petrovsky" sergej5...@yandex.com:Hello Thanks for the response! I already set both Agents to empty (also tried the username), doesn't make any difference. What I could figure out from the modem's SIP client is: #grep -ri agent /bin/vspa_sip HW_SIP_FormatUserAgentStrSIPPA_FunFillUserAgentHeadHW_VSPA_CheckUriAndUserAgentDomainHW_VSPA_CheckUserAgentDomainParseUserAgentMemCpSipUaDlgUAddUserAgentAndOrgnizationHeadersSipUaUtilAddUserAgentHeaderSipUserIeIniUserAgentHW_VSPA_CheckUserAgentDomain Failed.ulRet=0x%x.UserAgentUserAgentDomain Check Err!VoiceProfile.{%u}.Line.{%u}.URI=%s,aucUserAgentDomain=%s[VOIP] %s VoiceProfile[%lu] Line[%lu] UserAgentDomain Changed From %s to %s.User Agent:%sSipLmSetSoftConfigPara SIP_SOFT_CONFIG_ADD_USER_AGENT_FOR_ALL_UA_MSGS failed.ulRet=%uFeature: LOG/TRACE/STATISTICS/BACKUP/IPV4_SUPPORT/OPTIMIZE_APPMSG/MIB_STATISTICS/ETAG/ADD_USER_AGENT_HDR/32_BIT/User-AgentUser-Agent: bInsUserAgent: %uSipUaUtilAddUserAgentHeaderSipUaDlgUAddUserAgentAndOrgnizationHeadersAdd UserAgent and Organization header failedThe problem really is that it register and from that point you have no way knowing what goes wrong on the other side... Sergej 11.05.2013, 09:50, "Asghar Mohammad" asghar...@gmail.com:you can find in [general] section.useragent=asterisk ; Allows you to change the user agent string ; The default user agent string also contains the Asterisk ; version. If you don't want to expose this, change the ; useragent string.sdpsession=asterisk ; Allows you to change the SDP session name string, (s=) ; Like the useragent parameter, the default user agent string ; also contains the Asterisk version.;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)On Sat, May 11, 2013 at 5:16 AM, Nick Khamis sym...@gmail.com wrote:Sorry to chime in here, is it possible to change the "Server: Asterisk ", "s=Asterisk", and "o=" within sip.conf? What are the directives exactly please? Thanks in Advance, Nick. On 5/10/13, Asghar Mohammad asghar...@gmail.com wrote: hi, you can try to change sip user agent and sdp session s , owner in sip config same as your phone,s (modem). asterisk by default send user agent = asterisk version , s= asterisk , o= asterisk. some providers are not happy if they see "asterisk" word :) On Sat, May 11, 2013 at 12:27 AM, Sergej Petrovsky sergej5...@yandex.comwrote: Hi folks, What I trying to do here is exactly this: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html My provider given me a Huawei modem which have 2 phone jacks on it, but instead of using it I rather redirect my POTS number to my PBX. I ran into couple of bumps on the road but now it's "half-working". I extracted the SIP user, pass, server info from the modem and even managed to put my PBX into the same VLAN they use, on the exact same IP address like the modem but there is 1 problem: It seems this modem also sends some session ID to the ISP's sip server, something what Asterisk doesn't by default. So if I do this: 1, Let the modem register at the sip service (the phone number can be called and ringing out) 2, Disconnect the modem 3, Let the PBX connect to the SIP server 4, PBX accepts the calls 5, About 5-10 minutes later it stops doing it, when I call the number it shows busy (beep, beep, beep), no matter if I restart Asterisk or not it won't work anymore just if I do the same trick again I'm sure the remote SIP server breaks the voip channel or something, it does NOT drop me out tho, my PBX can register any time without problem but no packets will ever come forward me anymore. It's kind of hard to solve this from 1 side. There must be some solution for this. Please help! Thank You, Sergej -- _ -- 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:
Re: [asterisk-users] time zone setting in asterisk
On 05/12/13 12:18, Asghar Mohammad wrote: you can try to set usegmtime=no in cdr.conf I commented it out, as no is the default setting; but for some reason it was enabled on Gentoo installation. -- Joseph -- _ -- 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] time zone setting in asterisk
solved? On Sun, May 12, 2013 at 5:39 PM, Joseph syscon...@gmail.com wrote: On 05/12/13 12:18, Asghar Mohammad wrote: you can try to set usegmtime=no in cdr.conf I commented it out, as no is the default setting; but for some reason it was enabled on Gentoo installation. -- Joseph -- __**__**_ -- 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-**usershttp://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] Integrate Astreisk with SIP interface
On Sun, 12 May 2013, Asghar Mohammad wrote: for creating sip user add following in sip.conf [ivr_user] context=ivrcontext for ivr add following in extensions.conf [ivruser] [ivrcontext] -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
