Re: [asterisk-users] Sip Providers
Voipjet and Teliax seem to work for me. I used to recommend MPC (myphonecompany.com) but lately they have been having lots of dropped calls. They just told me that they got a new switch to handle more capacity. Will see how they work. But for now stick with what works. - Original Message - From: Alex Roston [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, July 07, 2007 7:11 PM Subject: [asterisk-users] Sip Providers Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex ___ --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] AudioCodes Gateway and Asterisk
Issue ended up being that the client was making the changes but he did not know that he needed to reset the box. Goto love when you are missing a bit of technical knowledge on the box ;) - Original Message - From: Dovid B [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 27, 2007 10:51 AM Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk Sent it to AudioCodes (in a text file). I will let you guys know what the issue was. - Original Message - From: Shanon Swafford [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, June 27, 2007 1:22 AM Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk When you see [ERROR] in the Message Log, either the MP firmware is buggy or the far end is sending something out of spec in the SIP Message. You'll need to upgrade to the latest MP firmware then report this to whomever you bought it from. Or fix the far end to send the message in spec or form that doesn't cause the [ERROR]. Also, do your supporter a favor and don't paste those logs directly into emails. The wrap makes them horrible to read and they can't send them on to Audiocodes like that. Put them in a text file which preserves the line length. Regards, Shanon http://www.abptech.com/support/qa/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, June 24, 2007 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk - Original Message - From: Shanon Swafford [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, June 21, 2007 6:27 PM Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk On 6/21/07, Dovid B [EMAIL PROTECTED] wrote: Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 Unsupported Media Type back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Are you sure, your asterisk supports G729? It isn't supported by default, you need additional modules or hardware cards for G729 support. If it is - what are you using for G729 - that might help to identify the problem. Regards, Atis If the AudioCodes is sending back that 415, the Message Log in the AudioCodes is invaluable. Set your debug level to 5/6 and watch it while you make test calls. Once you learn how to interpret this output, you'll be well on your way with AudioCodes. If G729 is active on the MP, but still giving back that error, G729 might not be in a profile if you are using them. Also, firmware that comes on the MPs is normally sorta buggy, ask your reseller for the latest version. http://www.abptech.com/support/faqs/ Regards, Shanon ABP Technology Shanon, The audiocodes were preftctly with other providers using G729. It's just having an issue with asterisk. Here is the output from the AudioCodes: Log is Activated 12d:23h:36m:17s ( lgr_flow)(828 ) Incoming SIP Message from XXX.XXX.XX.XXX:5060 [File: Line:-1] 12d:23h:36m:17s INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Record-Route: lt;sip:XXX.XXX.XX.XXX;ftag=as4a537e63;lr=ongt; Via: SIP/2.0/UDP XXX.XXX.XX.XXX;branch=z9hG4bK1db7.ecac6e86.0 Via: SIP/2.0/UDP XXX.XXX.XX.XXX:5060;branch=z9hG4bK1ef72aa9;rport=5060 From: 55560888 lt;sip:[EMAIL PROTECTED]gt;;tag=as4a537e63 To: lt;sip:[EMAIL PROTECTED]gt; Contact: lt;sip:[EMAIL PROTECTED]:5060gt; Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Enswitch Max-Forwards: 16 Date: Wed, 20 Jun 2007 19:44:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 490 v=0 o=root 29170 29170 IN IP4 XXX.XXX.XX.XXX s=session c=IN IP4 XXX.XXX.XX.XXX t=0 0 m=audio 14878 RTP/AVP 18 0 8 10 3 111 5 7 110 97 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no;mode-change-period=0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/80 12d:23h:36m:17s ( sip_stack)(830 ) ?? [WARNING] AcSIPParser: Unrecognized Header was detected at line: 12 [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(831 ) | | new GetNewSIPCall created - #8 [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(832 ) new AcSIPCallAPI created - #5 [File: Line:-1] 12d:23h:36m:17s ( lgr_stk_mngr)(833 ) Resource StackSession lt;#5gt; Allocated [File: Line:-1] 12d:23h:36m:17s ( lgr_flow)(834 ) | |(SIPTU#8)INVITE State:Idle() [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(835 ) SIPCall(#8) changes state from Idle to Invited [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(836 ) AcSIPParser: Problem in AcSIPCallAPI::ParseSDP [File: Line:-1] 12d:23h:36m:17s ( sip_stack)(837 )
Re: [asterisk-users] Sip Providers
Teliax has been great, VoipJet seems to be working vry hard to give outstanding reliability and fault notifications so I would recommend them as a backup. I like to have two account and failover for termination. -- Chris Mason Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670 International: (305) 704-7249 Fax: (815)301-9759 Yahoo IM only: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] installing * from source
- Original Message - From: Baji Panchumarti [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, July 07, 2007 7:02 PM Subject: [asterisk-users] installing * from source Just a quick listing of tested, and updated, steps from my notes. Enjoy ! http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html -baji. I was actually thinking of creating a script that you download and it preps your system for an asterisk install and it does everything for you. It can also have an option to run as a cron job and update nightly. The issue is that you cant just update some ones phone system if they are using it. So you would need like and email or sms sent to the user telling him to run the update script. What do others think of this idea ? ___ --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] Early Media Handling
Hi using php script and Asterisk manager I'm dialing numbers and once gets connected send to an exten in my dial plan that plays an automated message but some time without answering even it goes to my exten. How can I handle early media in Asterisk that is I want only when user answer the call it should goto my specified extension. my php script: $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: Originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, WaitTime: $strWaitTime\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); Please help thanks arun ___ --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] Session Border Controller time...
I was looking in to Nextone. What don't you like about them ? - Original Message - From: J. Oquendo [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 03, 2007 10:31 PM Subject: Re: [asterisk-users] Session Border Controller time... ___ --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] asterisk is not sip proxy
Edgar, On Sat, 7 Jul 2007, Edgar Guadamuz wrote: I know that Asterisk acts as a useragent endpoint, but my doubt is why exactly Asterisk could overload the call flow if the RTP voice stream goes from the caller to the called party. Asterisk is used commonly enough in carrier/transit applications precisely in this manner, and it is not necessarily untenable to do so. The distinction between a back-to-back user agent and a proxy is a rather formal one; the reason a B2BUA is not a proxy is more to do with the fact that even with re-INVITEs enabled, on the signaling level Asterisk builds out two distinct call legs with distinct GUIDs (Call-IDs) and cross-connects them, and intermediates the capability negotiation (codecs, packetisation delay, etc.) and establishment (media ports, etc.) heavily by virtue of that. A proxy, by contrast, is simply a SIP message router; it forwards SIP messages along a signaling path with whose fundamental protocol-level continuity it does not interfere. The practical reason to use something like [Open]SER over Asterisk for call placement has to do with the tasks for which it is intended to scale. A SIP proxy is specially optimised for handling large-scale call routing and its feature set and design pattern is largely built around that imperative. In the case of SER, it also has a number of advanced features that revolve around call routing and SIP header filtering / rewriting and other things one might typically expect from an element that plays a gateway role, and which are only partially present in Asterisk and not to the same degree. Both Asterisk and SER keep state information about SIP sessions and transactions, but Asterisk as an endpoint is likely to allocate far more memory and devote more programmatic hooks to each call leg on the premise that it is used as an endpoint or a feature server and not as a simple router. I am not an Asterisk developer, of course, so at best this is really a reasoned conjecture, and some people's mileage may vary. But if we take what I said as fact, SER comes out looking as a much more lean solution from the vantage point of pure message passing and routing. On the other hand, Asterisk has far more advanced and developed capabilities to interface with other applications and/or provide more high-level call management capabilities. A SIP proxy is really just about proxying, and not much else. That's why many VoIP billing/settlement/clearing platforms actually bounce calls off Asterisk, in order to take advantage of its CDR engine, etc. If you do not find those methodological considerations persuasive or they do not present at an issue at your volume and scale, however, they may ultimately be inconsequential to your application. And indeed, if you talk to many VoIP termination / origination providers, you will find that many of them use Asterisk in transit applications in some shape or form. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Sip Providers
- Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, July 08, 2007 11:03 AM Subject: Re: [asterisk-users] Sip Providers Teliax has been great, VoipJet seems to be working vry hard to give outstanding reliability and fault notifications so I would recommend them as a backup. I like to have two account and failover for termination. Why voipjet as a back up ? I use them as primary and if the call fails I send it over to Teliax. Why not give the client the cheaper price ? ___ --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] Session Border Controller time...
What does the NexTone run for ? - Original Message - From: Andy Brezinsky [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 03, 2007 8:17 PM Subject: Re: [asterisk-users] Session Border Controller time... We use NexTone for our SBC's on our network. We like: - 10,000 concurrent calls with media routing - SIP H.323 signaling with ability to take care of odd vendor specific issues - Basic routing engine allows you to create calling plans for individual end points - Limits by bandwidth or concurrent calls (or egress/ingress) for either discrete endpoints or via an iEdge group. - Easy GUI for those less tech savvy to do work on the machines. - Reasonable pricing on a per-port basis - Amazing Sales/Support teams. We've had some super funky requests we've thought about on a Friday night, they've got their teams together to walk us through every part of the configuration. Very knowledgeable and fun staff. (Seriously, best vendor support we've ever had, Hi Dan!) If you upgrade your SBC's to their RSM product you get basically a full Class 4 soft switch with a full LCR routing engine, reporting system and analytics engine. It's pretty powerful. Right now we're using just the SBC component and sending all ingress traffic to a egress trunk group (pointed to our OpenSER routers) but we're running a few thousand concurrent calls throught it. -- ~Andy Brezinsky On Tue, 2007-07-03 at 12:14 -0400, J. Oquendo wrote: Come on you carriers on the list... Give up the dibs what are you using and why? About to sledgehammer these SELECT * FROM GARBAGE WHERE SBC = 'nCite' Don't bother shooting me off Newport Networks stuff... Too pricey ___ --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 ___ --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] fail to load modules
Sorry for the late response but did you build the add-ons for asterisk ? - Original Message - From: clive.chan(Alpha Trilogies Networks) To: asterisk-users@lists.digium.com Sent: Thursday, June 28, 2007 12:20 PM Subject: [asterisk-users] fail to load modules Hi all, I am a bit out with the Asterisk 1.4.4, after I complied and installed the Asterisk and I got such error messages [Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose I got nothing error during installation of asterisk-addons-1.4.2 after I had change the Make file on the chan_ooh323.so.1.0.1. Tried; I tried to define noload to the chan_00h323.so and res_config_mysql.so, my asterisk start but give me others problems as bellowing... [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' could not be loaded. Can some one shares experience ?? -- ___ --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] installing * from source
On Sun, Jul 08, 2007 at 11:16:09AM +0300, Dovid B wrote: I was actually thinking of creating a script that you download and it preps your system for an asterisk install and it does everything for you. It can also have an option to run as a cron job and update nightly. The issue is that you cant just update some ones phone system if they are using it. So you would need like and email or sms sent to the user telling him to run the update script. What do others think of this idea ? How can you test the new version beforehand to know that it works well? Solution: prepare a package of Asterisk in your own repository. Build it on one system when the a new version is released. Test it. Update the repository after you've done testing. Updating a system nightly from a repository is a ewll-known problem with many solid solutions for various requirements. E.g: cron-apt on Debian. A such a script (in the pkg-voip repository) svn update # if there is a new version: update debian/changelog, and: debian/rules get-orig-source svn-buildpackage -rfakeroot -uc -us --svn-lintian -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Asterisk Help
Hi I need help in configuring a auto dialer system using Asterisk. I'm holding my customers number in MySQL want to fetch 10 numbers one time and dial if gets connected and answered by customer wants to play a sequence of message . Please help . I've tried here is my code to place calls but in this I see no of failure calls are more than 50%. so please advise. $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: Originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); thanks arun ___ --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] Zapata, Junghanns Card and a leading 0 on inbound calls
Hi, I'm using a Junghanns Quadbri ISDN card on some lines from the Austrian Telekom. Things are working, the only missing stuff is to add a 0 as a prefix to each incoming call, to make it possible to answer missed call lists. I'm using the 0 as the prefix for outside lines. I've experimented a little with the prefix settings in zapata.conf, but without success: pridialplan=national ;prilocaldialplan=unknown ;pridialplan = unknown ;prilocaldialplan = dynamic nationalprefix = 00 internationalprefix = 000 Is there a way to debug the ISDN messages coming on my D-channel, like it is shown with misdn? I would like to see the dialplan settings coming from the Network side. best regards -- Alexander Topolanek http://www.topolanek.at ___ --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] installing * from source
On 7/8/07, Dovid B wrote: [...] I was actually thinking of creating a script that you download and it preps your system for an asterisk install and it does everything for you. It can also have an option to run as a cron job and update nightly. The issue is that you cant just update some ones phone system if they are using it. So you would need like and email or sms sent to the user telling him to run the update script. What do others think of this idea ? Dovid, I am not sure about an update script due to reasons that Tzafrir and you already pointed out. But I think it would be GREAT to have an initial install script that just works, period ! For years installing/updating LAMP (apache, PHP MySQL on Linux) was a manual process, I read somewhere that Ubuntu now has a script that does the whole thing for you, and does it correctly. -baji. -- ___ --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] Zapata, Junghanns Card and a leading 0 on inbound calls
On Sun, Jul 08, 2007 at 03:22:33PM +0200, Alexander Topolanek wrote: Hi, I'm using a Junghanns Quadbri ISDN card on some lines from the Austrian Telekom. Things are working, the only missing stuff is to add a 0 as a prefix to each incoming call, to make it possible to answer missed call lists. I'm using the 0 as the prefix for outside lines. I've experimented a little with the prefix settings in zapata.conf, but without success: pridialplan=national ;prilocaldialplan=unknown ;pridialplan = unknown ;prilocaldialplan = dynamic nationalprefix = 00 internationalprefix = 000 Is there a way to debug the ISDN messages coming on my D-channel, like it is shown with misdn? I would like to see the dialplan settings coming from the Network side. bri debug span N bri intense debug span N -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] installing * from source
On Sun, Jul 08, 2007 at 10:05:37AM -0400, Baji Panchumarti wrote: On 7/8/07, Dovid B wrote: [...] I was actually thinking of creating a script that you download and it preps your system for an asterisk install and it does everything for you. It can also have an option to run as a cron job and update nightly. The issue is that you cant just update some ones phone system if they are using it. So you would need like and email or sms sent to the user telling him to run the update script. What do others think of this idea ? Dovid, I am not sure about an update script due to reasons that Tzafrir and you already pointed out. But I think it would be GREAT to have an initial install script that just works, period ! For years installing/updating LAMP (apache, PHP MySQL on Linux) was a manual process, I read somewhere that Ubuntu now has a script that does the whole thing for you, and does it correctly. Installing LAMP on Linux in the recent years has been something of the sort of: apt-get install apache php mysql (different package managers, different package names. Those are not even the actual package names in Debian). This provides you with an installation that you can easily upgrade on the next apache/php security hole. Any server distribution worth its salt has those (and let's not get into a distro fight in here) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Sip Providers
Alex Roston wrote: Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex I've been using www.axVoice.com for about 9 months now with great results. Quality is good, but communication seems to be best through email when dealing with them. Emails however, are returned very promptly. -- Warm Regards, Lee ___ --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] Zapata, Junghanns Card and a leading 0 on inbound calls
Am Sonntag, den 08.07.2007, 17:42 +0300 schrieb Tzafrir Cohen: bri debug span N bri intense debug span N thanks. Just for the records: pridialplan=national nationalprefix = 00 internationalprefix = 000 adds the required 0 in front of the number. But, a reload is not enough to reload these settings, asterisk has to be restarted for that. BTW, is it possible to set these functions per group? best regards -- Alexander Topolanek http://www.topolanek.at ___ --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] asterisk is not sip proxy
Alex Balashov wrote: The distinction between a back-to-back user agent and a proxy is a rather formal one; (. . . . ) I suggest the essence of this mail be distilled and put into a FAQ somewhere, if it isn't already. Thanks. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] fail to load modules
Hi All Ive got the same message after installing asterisk addons [res_convert.so]Jul 8 20:51:10 WARNING[4685]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_convert.so: undefined symbol: ast_module_unregister Jul 8 20:51:10 WARNING[4685]: loader.c:554 load_modules: Loading module res_convert.so failed! Your help please _ De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Dovid B Envoyé : dimanche 8 juillet 2007 10:31 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] fail to load modules Sorry for the late response but did you build the add-ons for asterisk ? - Original Message - From: clive.chan(Alpha mailto:[EMAIL PROTECTED] Trilogies Networks) To: asterisk-users@lists.digium.com Sent: Thursday, June 28, 2007 12:20 PM Subject: [asterisk-users] fail to load modules Hi all, I am a bit out with the Asterisk 1.4.4, after I complied and installed the Asterisk and I got such error messages [Jun 28 16:56:19] WARNING[28625] res_smdi.c: No SMDI interfaces are available to listen on, not starting SDMI listener. [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 28 16:56:19] WARNING[28625] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose I got nothing error during installation of asterisk-addons-1.4.2 after I had change the Make file on the chan_ooh323.so.1.0.1. Tried; I tried to define noload to the chan_00h323.so and res_config_mysql.so, my asterisk start but give me others problems as bellowing... [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'cdr_addon_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_addon_sql_mysql.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'format_mp3.so' could not be loaded. [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' did not register itself during load [Jun 28 17:03:12] WARNING[28637] loader.c: Module 'app_saycountpl.so' could not be loaded. Can some one shares experience ?? _ ___ --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] North American voice BRI - Informal survey
The other thing that I was thinking is that I prefer PRI to analog so much that I even if it cost a hundred bucks more a month, it's still attractive to me. All that tends to support our contention that there should be a market for NA BRI support. You'd think many installations would benefit. Don't forget that BRI is quite different from PRI in various ways. For example, the handling of phone numbers is usually substantially different, you cannot generally set outbound Caller-ID as one example. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] Zapata, Junghanns Card and a leading 0 on inbound calls
On Sun, Jul 08, 2007 at 06:33:01PM +0200, Alexander Topolanek wrote: Am Sonntag, den 08.07.2007, 17:42 +0300 schrieb Tzafrir Cohen: bri debug span N bri intense debug span N thanks. Just for the records: pridialplan=national nationalprefix = 00 internationalprefix = 000 adds the required 0 in front of the number. But, a reload is not enough to reload these settings, asterisk has to be restarted for that. BTW, is it possible to set these functions per group? Not per-group, but I believe you can set them per-chanel. A group is an arbitrary set of channels. pridialplan=unknown channel =1-2 pridialplan=national nationalprefix = 00 internationalprefix = 000 channel =4-5 Or something similar. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Unable to install Asterisk Now Beta 6
Hi Bart,
Thank you very much for your message.
I don't think the problem is with dual proc systems, because
the install failed as well on an old Penitum 4, mono-proc.
Thnk for giving that link to elastix.org, seems to be a very
nice option, and the last release is dated of early July.
I really wonder if someone has been able to install Asterisk
Now Beta 6 using the iso file published on their website...
Thanks.
-- Debut du message initial ---
De : [EMAIL PROTECTED]
A : Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Copies :
Date : Sat, 07 Jul 2007 08:45:19 -0700
Objet : Re: [asterisk-users] Unable to install Asterisk Now
Beta 6
I don't believe AsteriskNow will install on a dual processor
system. I
had this same problem - installing on single process MB went OK
I don't know how to fix, so went with elastx.org and
adminsparadise.com
packages, both seemed to be OK - can't decide which one to
keep - the
last choice, maybe should be first choice is trixbox - it's
the best
supported package out there for the newbie - but does not
support
Hylafax and asterisk 1.4 (yet) like the other two. They say
it's coming :)
Bart
mtest001 wrote:
Hi everybody !
I'm desperately trying to install AsteriskNow Beta 6. I
downloaded the iso file (version x86 32 bits) and burned it,
then I tried on three different computers (from an old Pentium
4 to a brand new HP DL380 2xDual Core) and each time I got the
same error...
Shortly after the installation begins, after the probing of
hardware component, the installer stops with the following
message :
Quote:
Running Anaconda [...]
file /usr/bin/anaconda, line 316, in ?
if (os.path.exists('isys')):
AttributeError: 'module' object has no attribute 'path'
...and then ask to reboot.
Am I the only one to have this error ? I burned two CDs and
tried on three computers ... no luck. It seems to me that
there's something wrong with this iso... Sad
Appreciate your help !
Btw I've got a question... I'm new to Asterisk and until now I
only configured it by editing the text files. I like to have
in my dialplan a macro that sends the caller to the voicemail
if the extension called is not available or does not answer in
15 seconds.
Is it possible to configure such a rule with the GUI of
Asterisk Now ? Is it possible to make it generic for each and
every extension ?
Thank you for your help.
Créez votre adresse électronique [EMAIL PROTECTED]
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Re: [asterisk-users] North American voice BRI - Informal survey
Has anyone got a PBX with spare BRI ports in it? Maybe that's a cheap way to get started. We could just hook a box up to that and work out some of the early stage stuff. I know that people with Polycom (and other) video/teleconferencing equipment often have BRI cards in their Nortel PBX or Avaya gear. Well, I've got a PBX (Asterisk) with some spare BRI ports (the previously described Adtran 550). I have one port that is definitely free. I might have another that could be freed most of the time if the cause was sufficient. I have no BRI cards. I do have some other ISDN gadgets. I'm willing to consider placing a small server at the disposal of a developer or something like that, if it'll lead towards better support, but what card and who provides it is up in the air (I am probably not in sufficient need to justify footing the bill for a several hundred dollar card, though I'd be fine popping for a $50 card). If this was sufficiently useful and there was actually forward progress, I might be willing to do more, like provide additional BRI ports on the Adtran, or maybe even short term access to a real US BRI line, or even fund a card or two. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] North American voice BRI - Informal survey
Maybe, but it will probably mean writing another driver just to provide telco-side signalling -- or is it the same on each end? No, I'm reasonably positive that there's a well defined user and network side. What's the deal with PRI cards? Can you run those back-to-back? Yes, usually. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] Not able to find the file zaptel.conf after compiling asterisk and zaptel
I posted this response to bilal ghayyad's inquiry a few days ago, but it hasn't reached everyone on the list yet (I copied bilal ghayyad so he should have received the information right away). Your post took almost a week to reach me. I have emailed the people identified as responsible for the list about delayed distribution, but have had no response from them. Can you have someone look into why this is happening? --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Bryant Sent: Sunday, July 01, 2007 3:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Not able to find the file zaptel.conf after compiling asterisk and zaptel bilal ghayyad wrote: I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? zaptel.conf is located at /etc/zaptel.conf, not /etc/asterisk/zaptel.conf. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] NAT
On 6/5/07, Tom Rymes [EMAIL PROTECTED] wrote: On Jun 5, 2007, at 9:46 AM, Cosmin Prund wrote: -Original Message- From: [EMAIL PROTECTED] [mailto: asterisk-users- [EMAIL PROTECTED] On Behalf Of Henry Cobb Sent: Tuesday, June 05, 2007 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NAT On 6/5/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote: Hi All!! I have my asterisk working in my house (working with mandriva 2007 and asterisk 1.4 svn). I´ve looking on the net but i couldn´t find the way of making work my asterisk in a real enviroment. Seems that the problem of NAT is a big problem. How can I sort out this, a mean crossing the NAT and having asterisk connected? If you want to receive calls and not just place them and you have a broadband connection with a dynamic IP then your server must register with the VoIP provider and I suggest using IAX with the proper UDP port assigned to your Atrisk server. -HJC NAT is not that big of a problem, not anymore. Do a NAT search on http://www.voip-info.org - it'll get you started (got me started at least) -- Cosmin Prund Specifically, you need to set the following in sip.conf (if applicable) nat= localnet= externip= externhost= You also need to configure your router to forward port 5060 and ports 1-2 to your asterisk server. You make it sound very easy :-) I've got a host connected to the internet with eth1 and to an internal LAN with eth0. The host runs asterisk. The internal LAN contains a number of SIP phones. eth0 = 192.168.254.254 (network 192.168.254.0/24) eth1 = internet IP-address I've set externhost to the dyndns name I've registered. When I do a lookup, this name returns the same IP-address as the one on eth1. I've got a DID, and when I dial that number from my cell, the phones ring in my home. When I pick up the phone, audio only goes one way (from my home phone (behind the NAT) to the DID) audio the other way (from the DID to my home phone behind the NAT) is missing, due to NAT. It figured because my asterisk server tells the DID to send the audio to the IP-address of my SIP phone on the internal network (192.168.254.105). I fired up wireshark and captured the packets. What I want to accomplish: - calls within the LAN are re-invited (RTP goes from endpoint to endpoint) - asterisk detects when a call is going beyond the local LAN (over the NAT), and then stays in the middle. I'm wondering if this is hard to do and how I'm supposed to configure this. regards, Stefan ___ --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] North American voice BRI - Informal survey
Quoting Joe Greco [EMAIL PROTECTED]: I've been in contact with a guy on ebay selling 4 port bri cards who supplies a patched version of the driver that works with north american signalling. my suggestion would be for someone to just take the plunge and see if it actually works properly and then provide the rest of us some feedback. There is absolutely no point in reinventing the wheel when the guy has reasonably priced cards and software which already works. Has anyone got a PBX with spare BRI ports in it? Maybe that's a cheap way to get started. We could just hook a box up to that and work out some of the early stage stuff. I know that people with Polycom (and other) video/teleconferencing equipment often have BRI cards in their Nortel PBX or Avaya gear. Well, I've got a PBX (Asterisk) with some spare BRI ports (the previously described Adtran 550). I have one port that is definitely free. I might have another that could be freed most of the time if the cause was sufficient. I have no BRI cards. I do have some other ISDN gadgets. I'm willing to consider placing a small server at the disposal of a developer or something like that, if it'll lead towards better support, but what card and who provides it is up in the air (I am probably not in sufficient need to justify footing the bill for a several hundred dollar card, though I'd be fine popping for a $50 card). If this was sufficiently useful and there was actually forward progress, I might be willing to do more, like provide additional BRI ports on the Adtran, or maybe even short term access to a real US BRI line, or even fund a card or two. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] installing * from source
Have you also consider adding adding the uBuntu steps in addition to CentOS? -E On 7/8/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Jul 08, 2007 at 10:05:37AM -0400, Baji Panchumarti wrote: On 7/8/07, Dovid B wrote: [...] I was actually thinking of creating a script that you download and it preps your system for an asterisk install and it does everything for you. It can also have an option to run as a cron job and update nightly. The issue is that you cant just update some ones phone system if they are using it. So you would need like and email or sms sent to the user telling him to run the update script. What do others think of this idea ? Dovid, I am not sure about an update script due to reasons that Tzafrir and you already pointed out. But I think it would be GREAT to have an initial install script that just works, period ! For years installing/updating LAMP (apache, PHP MySQL on Linux) was a manual process, I read somewhere that Ubuntu now has a script that does the whole thing for you, and does it correctly. Installing LAMP on Linux in the recent years has been something of the sort of: apt-get install apache php mysql (different package managers, different package names. Those are not even the actual package names in Debian). This provides you with an installation that you can easily upgrade on the next apache/php security hole. Any server distribution worth its salt has those (and let's not get into a distro fight in here) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Asterisk and Mitel 3300 ICP
Good day everyone, I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and from extensions on both sides are completing successfully (details on config coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel 3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN calls through it successfully? Here is an extract of the log on Asterisk whenever I try to call PSTN through 3300ICP, in this case, Extension 2540 on Asterisk called 2345678, 9 is a leading digit - Jul 7 16:48:07 DEBUG[6860] chan_sip.c: Outgoing Call for 92345678 Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Called Mitel3300ICP/92345678 Jul 7 16:48:07 DEBUG[6860] channel.c: Prodding channel 'SIP/2540-b7904a98' Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Started music on hold, class '24', on SIP/2540-b7904a98 Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 160 sample intervals Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Acked pending invite 102 Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Stopping retransmission on ' [EMAIL PROTECTED]' of Request 102: Match Found Jul 7 16:48:07 DEBUG[6860] channel.c: Generator got voice, switching to phase locked mode Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 0 sample intervals Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 103: Found Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Acked pending invite 103 Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Stopping retransmission on ' [EMAIL PROTECTED]' of Request 103: Match Found Jul 7 16:48:08 WARNING[6270] chan_sip.c: Forbidden - wrong password on authentication for INVITE to 'Tester sip:[EMAIL PROTECTED];tag=as07fef065' Jul 7 16:48:08 VERBOSE[6860] logger.c: -- SIP/Mitel3300ICP-0832de50 is circuit-busy ___ --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] Sip trunk between Asterisk and Mitel 3300 ICP
hallo everyone, fyi ... working SIP Trunk configuration between Asterisk and Mitel 3300 ICP attached. let's refine further, please test and share your feedback, regards, Joseph Okoegwale Abuja, Nigeria SIP Trunk between Asterisk and Mitel 3300 ICP PBX Source/Credit - Timo Sariwating http://www.sundance-communications.com/forum/ultimatebb.php?/ubb/get_topic/f/6/t/000558.html#00 Mitel 3300ICP = 192.168.1.2 Number range = 5xx Trixbox = 192.168.1.101 Number range = 25xx On the Mitel 3300ICP - 1. Network Element Assignment - create a network element for the local switch - create a network element for each SIP peer, gateway, or Service Provider - create a network element for the Outbound Proxy if one exists in your network Ensure there is a local element for the Mitel. if there is none, create it. Create a network element for the Asterisk box Name - Asterisk Type = Other FQDN or IP adress = 192.168.1.101 (Asterisk IP Address) SIP Peer = selected external FQDN or IP = 192.168.1.101 (Asterisk IP Address) SIP registrar FQDN or IP = 192.168.1.101 (Asterisk IP Address) Transport = UDP and Port = 5060 for all 2. System IP Port Assignment Change the SIP UDP, TCP, or TLS port number if it is different from the default value. SIP UDP = 5060 SIP TCP = 5060 SIP TLS = 5061 3. DID Ranges for CPN Substitution To set up the CPN Substitution table for outbound calls, enter a DID number or a range of DID numbers assigned in the system. Then enter the corresponding CPN substitution number that will be delivered for that range e.g Index = 10 DID Range = 500-599 CPN Substitution = 5XX 4. Create a SIP Peer for Asterisk - Use SIP Peer Profile Form SIP peer profile label = Asterisk Local Account registration username = 150 (an extension that would be used for authentication, should match in Asterisk) Adress type = IP adress : 192.168.1.2 (ip address of 3300 ICP) Authentication username = 150 (an extension that would be used for authentication, should match in Asterisk) password en confirm password = abcd (password set on extension, should match in Asterisk) Authentication = Challenge-based Authentication Outgoing DID Ranges: select index 10 (select matching index if Calling Party Number Substitution was configured) 5. Optional - SIP Peer Profile Assignment for Incoming DID To associate a range of telephone numbers assigned by a SIP Service Provider to a particular SIP Peer, enter the required information in this form. 6. Trunk Service Assignment: Configure the trunk as non-dial in or dial-in: - update the Non-Dial-In Trunks Answer Point field for the incoming calls. - strip the number of leading digits in Dial-In Trunks Incoming Digit Modification Absorb field - add the appropriate number of digits in Dial-In Trunks Incoming Digit Modification Insert field. Trunk service number = 10 (based on my situation) class of service = 64 (Enter the COS number that defines the required options for the trunk) class of restriction = 64 (Enter the COR number for the trunk. This COR number must not have been assigned to a station (mandatory field)) trunk label = Asterisk trunk Dial in Trunks Incoming Digit absorb = 0 (you can use this to do leading digits absorption etc) 7. Class of Service Options Assignment Enable the Public Network Access via DPNSS field in the class of service for all devices that make outgoing calls through SIP trunks, PRI trunks, LS trunks, and so forth that are connected to SIP Trunks. 8. Route Assignment Complete the following fields in this form: - select SIP Trunk from the pull-down list in Routing Medium. - select a SIP Peer Profile label from the SIP Peer Profile pull-down list. - enter a Class of Restriction group number in COR Group Number (this determines which extensions *cannot* access this trunk, I am using a COR that permits all Mitel extensions to access the Sip trunk and therefore call Asterisk users successfully) - enter any required digits in Digits Before Outpulsing. (If this field is left blank, digits will be sent out as Enbloc.) Route number = 10 Routing medium = SIP Trunk Trunk group number = empty SIP Peer profile = Asterisk Route Type = PSTN access via DPNSS - ARS Digits dialed Assignment: Digits Dialed = 2 (2 is the first digit of my Asterisk extensions Numbers to follow = 3 (5xx follow = 3 digits) Termination Type = route Termination number = 10 (route number created above) Make sure to enable Public Network Access via DPNSS in the SIP trunk COS. On the Trixbox - - Create a SIP trunk: Outgoing Trunk name = Mitel PEER Details: allow=ulaw auth=md5 context=from-pstn host=192.168.1.2 insecure=very nat=no secret=abcd type=peer username=150 - Create a SIP Extension: Display name = Mitel 3300ICP Device options: secret = abcd canreinvite = no context = from-internal host = dynamic type = peer nat = no port = 5060 dial = SIP/150 - Create an outbound route: Route name = Mitel3300ICP Dial patterns = 5XX Trunk Sequence = 0 SIP/Mitel - Create inbound routes
Re: [asterisk-users] voicemail.conf serveremail
Patrick Pfeifer wrote: Hello, I was wondering if there is a way to change the From address (not just the Return-Path) for voicemail notification emails in Asterisk. It looks like the serveremail directive in voicemail.conf just changes the Return-Path. I'm looking for something analogous to the -r option in mailx, for example. I need this since the mail server I'm using requires the sender to be on the system. Any advice would be appreciated. Have you considered an alternative mailer, like ssmtp? That's what we use -- and we just define the From: address in the configuration for ssmtp. -Stephen- ___ --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] fail to load modules
Hi Dovid, My problem solved, advice from Bryant. Thank you for your concern. :-) ___ --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
