Re: [asterisk-users] HA Asterisk
On 04/30/2011 02:13 AM, RAJNIKANT VANZA wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. Requests for additional details are a lot more persuasive when conveyed in a semi-literate manner. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] HA Asterisk
Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan kaushalshri...@gmail.comwrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- 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] HA Asterisk
On Sat, Apr 30, 2011 at 11:47 AM, Alex Balashov abalas...@evaristesys.comwrote: On 04/30/2011 02:13 AM, RAJNIKANT VANZA wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. Requests for additional details are a lot more persuasive when conveyed in a semi-literate manner. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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 Hi, Can someone please answer the rest of the questions in my earlier post to this Mailing List.? Regards, Kaushal -- _ -- 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] Best modem for chan_datacard
Huawei e180, K3715 are good to play around. Both voice and SMS are supported. On Fri, Apr 29, 2011 at 2:47 AM, Tiago Geada tiago.ge...@gmail.com wrote: I used succesfully huawei E1550 On 24 April 2011 16:45, Dovid Bender asteriskus...@dovid.net wrote: Hi List, I am looking to play around with chan_datacard. Any advice on the best device to test with (that I can find on eBay) ? Regards, Dovid -- _ -- 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] odbc error - server is gone
check your odbc connection with isql best On Fri, Apr 29, 2011 at 9:33 PM, Warren Selby wcse...@selbytech.com wrote: You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I believe 1.4.41 is current) and see if your issue has been resolved. Thanks, --Warren Selby, dCAP On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Yes I have it there, here the content of the file: i think the code is buggy, here is a comment from the function which generated the error (ast_odbc_smart_execute in res_odbc.c line 155 ) /* This is a really bad method of trying to correct a dead connection. It * only ever really worked with MySQL. It will not work with any other * database, since most databases prepare their statements on the server, * and if you disconnect, you invalidate the statement handle. Hence, if * you disconnect, you're going to fail anyway, whether you try to execute * a second time or not. */ This function is used all over. On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan sherwood.mcgo...@gmail.com sherwood.mcgo...@gmail.com wrote: On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.com rizwanhas...@gmail.com wrote: Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemailand herehttp://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage . I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for mysql on the server. I successfully completed the conversion of a lot of voicemail users into db yesterday. But today on the CLI thsi error was showing; [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users do you have sanitysql = select 1 configured in res_odbc.ini? -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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] SIP bad request
may be the ip phone has the problem, try reset as factory On Fri, Apr 29, 2011 at 8:03 PM, Mike l...@net-wall.com wrote: What I am looking for? Here is a snippet, with some info obfuscated. I can see the bad request, but why there is such a message isn’t obvious. --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Allow-Events: talk,hold,conference Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 - --- (11 headers 0 lines) --- --- SIP read from UDP:23.23.23.23:23725 --- SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 66.66.66.66:5060;branch=z9hG4bK742ee2b8;rport From: JOHN SMITH sip:55@66.66.66.66;tag=as40e0c5af To: user sip:user@192.168.1.90:5060;tag=372AEEC-62912E9F CSeq: 102 INVITE Call-ID: 49975a6153b9213972edbdf263186863@66.66.66.66 Contact: sip:user@192.168.1.90:5060 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: fr-fr,fr;q=0.9,en;q=0.8 Content-Length: 0 *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *??? ? *Sent:* Friday, April 29, 2011 10:49 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] SIP bad request Try to look in 'sip set debug peer user'. On 29.04.2011 18:10, Mike wrote: Hi, I have been getting reports phones ringing only a tiny moment and then going to voicemail. CLI output shows: -- SIP/user-0006fcdd is ringing -- Got SIP response 400 Bad Request back from 23.23.23.23 -- SIP/user-0006fcdd is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Which does explain it. How can I find the root cause of “bad request”? Call-limit is very high for this sip user, so I`m not reaching that limit for sure. Mike -- _ -- 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] AGI WAIT FOR DIGIT - key press BEFORE command
Dear try phpagi. it has a lot of useful functions. in this scenario you will lose your digit, set a check point between each digit gathering best On Wed, Apr 27, 2011 at 6:17 PM, David asterisk@spam.lublink.netwrote: Hi, Consider the following situation : SIP/asterisk-001dAGI Rx WAIT FOR DIGIT 3000 SIP/asterisk-001dAGI Tx 200 result=48 SIP/asterisk-001dAGI Rx WAIT FOR DIGIT 3000 SIP/asterisk-001dAGI Tx 200 result=48 SIP/asterisk-001dAGI Rx WAIT FOR DIGIT 3000 SIP/asterisk-001dAGI Tx 200 result=48 SIP/asterisk-001dAGI Rx WAIT FOR DIGIT 3000 What happens if the user enters a digit between the 200 result= and the next WAIT FOR DIGIT? Will the next WAIT FOR DIGIT catch the digit? Is the digit lost? How can I insure I don't lose the digit ? I am calling one digit at a time because I want to validate the user's entry at each key press. Thanks, David -- _ -- 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] HA Asterisk
Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf , but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- 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] HA Asterisk
You can't do PRI failover while using internal PRI cards. To do so you need a standalone PRI box a good one i use often is foneBridge from Redfone. U can use foneBridge as follow: PSTN(Tdm) - FoneBridge(TDMoE)- Asterisk . Adolphe Cher-aime From my Iphone On Apr 30, 2011, at 9:09 AM, Satish Patel satish...@hotmail.com wrote: Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf , but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- 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] HA Asterisk
On 04/30/2011 10:20 AM, Adolphe Cher-aime wrote: You can't do PRI failover while using internal PRI cards. To do so you need a standalone PRI box a good one i use often is foneBridge from Redfone. U can use foneBridge as follow Hi, You can do a PRI failover with Dataprobe switches. Use the monit daemon to check for red alarms in syslog, then shutdown asterisk, then shutdown the PRI, the backup PRI is auto switched through the Dataprobe. Adrian Serafini -- _ -- 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] HA Asterisk
Use simple RJ45 (8 wire) A-B switched controllable by serial port, and use HAAST to throw the A-B switch to reroute the PRI. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Adolphe Cher-aime [achera...@gmail.com] Sent: Saturday, April 30, 2011 10:20 AM To: Asterisk Users List Cc: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk You can't do PRI failover while using internal PRI cards. To do so you need a standalone PRI box a good one i use often is foneBridge from Redfone. U can use foneBridge as follow: PSTN(Tdm) - FoneBridge(TDMoE)- Asterisk . Adolphe Cher-aime From my Iphone On Apr 30, 2011, at 9:09 AM, Satish Patel satish...@hotmail.commailto:satish...@hotmail.com wrote: Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA mailto:rajniva...@gmail.comrajniva...@gmail.commailto:rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan mailto:kaushalshri...@gmail.commailto:kaushalshri...@gmail.comkaushalshri...@gmail.commailto:kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf http://www.asterisk.org/asterisk/scf http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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] dial multiple extensions
Hello, I've got a problem with something I'm doing and can't seem to figure it out. I've tried different suggestions I've found on voip-info.org as well as other sites but nothing I do seems to work. I've got an older Digium TDM400P. The FXO daughter card is connected to my POTS line and the FXS daughter card is connected to a TDM phone. I also have multiple SIP extensions. My desire is to ring all the internal extensions (the TDM and SIP extensions) on an inbound call and send the call to whichever extension picks up first. This seems to be working just fine if the extension that picks up is one of the SIP phones. On the other hand, if the extension that picks up is the one off the FXS port, then the SIP phones continue to ring and the dial plan continues to execute even though the caller on the FXO port has been connected to the phone on the FXS port. Inbound calls are sent to extension 3100, which looks like this: exten = 3100,1,Dial(SIP/3105SIP/3106SIP/3108dahdi/1,20,tr) exten = 3100,n,Voicemail(3100) exten = 3100,n(end),Hangup() Like I said, if I pick up on one of the SIP extensions, it seems to do exactly as I expect. If I pick up on dahdi/1, however, the SIP phones continue to ring and the FXO and FXS ports are connected and passed into voicemail. I'm running on Debian Squeeze/6.0.1 and am running the stock Asterisk 1.6.2.9-2+squeeze2 package. If anyone has some suggestions, I'd be happy to hear them. Thanks! Roy -- _ -- 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] dial multiple extensions
I don't think this will solve your issue but I would say remove r option in dial command I had same issue with iax and sip phone and I solved with that r option Hope it will help you. -- Sent from my iPhone On Apr 30, 2011, at 1:15 PM, Roy Kidder rkid...@rkidder.com wrote: Hello, I've got a problem with something I'm doing and can't seem to figure it out. I've tried different suggestions I've found on voip-info.org as well as other sites but nothing I do seems to work. I've got an older Digium TDM400P. The FXO daughter card is connected to my POTS line and the FXS daughter card is connected to a TDM phone. I also have multiple SIP extensions. My desire is to ring all the internal extensions (the TDM and SIP extensions) on an inbound call and send the call to whichever extension picks up first. This seems to be working just fine if the extension that picks up is one of the SIP phones. On the other hand, if the extension that picks up is the one off the FXS port, then the SIP phones continue to ring and the dial plan continues to execute even though the caller on the FXO port has been connected to the phone on the FXS port. Inbound calls are sent to extension 3100, which looks like this: exten = 3100,1,Dial(SIP/3105SIP/3106SIP/3108dahdi/1,20,tr) exten = 3100,n,Voicemail(3100) exten = 3100,n(end),Hangup() Like I said, if I pick up on one of the SIP extensions, it seems to do exactly as I expect. If I pick up on dahdi/1, however, the SIP phones continue to ring and the FXO and FXS ports are connected and passed into voicemail. I'm running on Debian Squeeze/6.0.1 and am running the stock Asterisk 1.6.2.9-2+squeeze2 package. If anyone has some suggestions, I'd be happy to hear them. Thanks! Roy -- _ -- 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] HA Asterisk
Your idea sounds good to me could you please explain more or point me to any document of website. -- Sent from my iPhone On Apr 30, 2011, at 12:09 PM, Michelle Dupuis mdup...@ocg.ca wrote: Use simple RJ45 (8 wire) A-B switched controllable by serial port, and use HAAST to throw the A-B switch to reroute the PRI. From: asterisk-users-boun...@lists.digium.com [asterisk-users- boun...@lists.digium.com] On Behalf Of Adolphe Cher-aime [achera...@gmail.com] Sent: Saturday, April 30, 2011 10:20 AM To: Asterisk Users List Cc: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk You can't do PRI failover while using internal PRI cards. To do so you need a standalone PRI box a good one i use often is foneBridge from Redfone. U can use foneBridge as follow: PSTN(Tdm) - FoneBridge(TDMoE)- Asterisk . Adolphe Cher-aime From my Iphone On Apr 30, 2011, at 9:09 AM, Satish Patel satish...@hotmail.commailto:satish...@hotmail.com wrote: Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA mailto:rajniva...@gmail.com rajniva...@gmail.commailto:rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan mailto:kaushalshri...@gmail.com mailto:kaushalshri...@gmail.comkaushalshri...@gmail.commailto:kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf http://www.asterisk.org/asterisk/scf http://www.asterisk.org/asterisk/scf , but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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
[asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend
Good Evening, I'm setting up an Internet Radio website with call-in functionality, and need to know the kinds of FOSS tools I should install to get the job done. Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png Call protocol: [Producer calls in] [Host calls in] [Guest calls in]-[Screened by Producer, if accepted, conferenced into host] On the website they'll need to be able to call in (mic input grabbed), and listen in (without calling in). I've been suggested many things, including Skype, IceCAST and [currently the most promising] Asterisk+Red5+Red5Phone. Are there any better ways of doing this, and if not, how do I setup asterisk for the above task? Thanks for all suggestions, Alec Taylor -- _ -- 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] HA Asterisk
There are lots out there, but here's the result of a quick search... http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html and the software to trigger the switch: www.generationd.com From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel [satish...@hotmail.com] Sent: Saturday, April 30, 2011 3:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk Your idea sounds good to me could you please explain more or point me to any document of website. -- Sent from my iPhone On Apr 30, 2011, at 12:09 PM, Michelle Dupuis mdup...@ocg.ca wrote: Use simple RJ45 (8 wire) A-B switched controllable by serial port, and use HAAST to throw the A-B switch to reroute the PRI. From: asterisk-users-boun...@lists.digium.com [asterisk-users- boun...@lists.digium.com] On Behalf Of Adolphe Cher-aime [achera...@gmail.com] Sent: Saturday, April 30, 2011 10:20 AM To: Asterisk Users List Cc: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk You can't do PRI failover while using internal PRI cards. To do so you need a standalone PRI box a good one i use often is foneBridge from Redfone. U can use foneBridge as follow: PSTN(Tdm) - FoneBridge(TDMoE)- Asterisk . Adolphe Cher-aime From my Iphone On Apr 30, 2011, at 9:09 AM, Satish Patel satish...@hotmail.commailto:satish...@hotmail.com wrote: Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA mailto:rajniva...@gmail.com rajniva...@gmail.commailto:rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan mailto:kaushalshri...@gmail.com mailto:kaushalshri...@gmail.comkaushalshri...@gmail.commailto:kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf http://www.asterisk.org/asterisk/scf http://www.asterisk.org/asterisk/scf , but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis mdup...@ocg.ca wrote: There are lots out there, but here's the result of a quick search... http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html and the software to trigger the switch: www.generationd.com Hi Michelle So what i understand is that the Single PRI Line from telco is connected to RJ45 (8 wire) A-B switched controllable by serial port and then there will be two patch cord from the A-B switch which will be connected to the 2 Asterisk Box containing PRI Card on each box. Please let me know if i am understanding you correctly or if you can help me with Network Diagram that would be really helpful. Also I have 8 PRI in my setup. How it would fit in this setup. The reason being we need to have atleast 320 Outbound Calls per min if i have 8 PRI Lines for our Voice Application. Regards, Kaushal -- _ -- 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] HA Asterisk
Yes that's it - one PRI line in, 2 out (one to the PRI card in each server). If you have lots of PRI lines, you may want to consider a dedicated PRI-to-SIP appliance.. From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan [kaushalshri...@gmail.com] Sent: Saturday, April 30, 2011 11:03 PM To: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: There are lots out there, but here's the result of a quick search... http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html and the software to trigger the switch: www.generationd.comhttp://www.generationd.com Hi Michelle So what i understand is that the Single PRI Line from telco is connected to RJ45 (8 wire) A-B switched controllable by serial port and then there will be two patch cord from the A-B switch which will be connected to the 2 Asterisk Box containing PRI Card on each box. Please let me know if i am understanding you correctly or if you can help me with Network Diagram that would be really helpful. Also I have 8 PRI in my setup. How it would fit in this setup. The reason being we need to have atleast 320 Outbound Calls per min if i have 8 PRI Lines for our Voice Application. Regards, Kaushal -- _ -- 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] HA Asterisk
On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis mdup...@ocg.ca wrote: Yes that's it - one PRI line in, 2 out (one to the PRI card in each server). If you have lots of PRI lines, you may want to consider a dedicated PRI-to-SIP appliance.. Hi, Thanks a Lot Michelle, Also please let me know the model/make for dedicated PRI-to-SIP appliance. Would appreciate if you can share the details along with the Network Diagram in case of 8 PRI Lines. Much appreciated. Regards, Kaushal From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan [ kaushalshri...@gmail.com] Sent: Saturday, April 30, 2011 11:03 PM To: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis mdup...@ocg.camailto: mdup...@ocg.ca wrote: There are lots out there, but here's the result of a quick search... http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html and the software to trigger the switch: www.generationd.comhttp://www.generationd.com Hi Michelle So what i understand is that the Single PRI Line from telco is connected to RJ45 (8 wire) A-B switched controllable by serial port and then there will be two patch cord from the A-B switch which will be connected to the 2 Asterisk Box containing PRI Card on each box. Please let me know if i am understanding you correctly or if you can help me with Network Diagram that would be really helpful. Also I have 8 PRI in my setup. How it would fit in this setup. The reason being we need to have atleast 320 Outbound Calls per min if i have 8 PRI Lines for our Voice Application. Regards, Kaushal -- _ -- 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
