Hi Bryant please use SRV records whenever a failover is required. This your hosting provider should be able to give to you and always try and avoid using direct ip calls.
Potentially you could create two A records: db1.yoursite.com db2.yoursite.com Then create a SRV record with priority on db1 over db2. This way you can add n-amount of additional failovers. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: Wednesday, September 28, 2011 6:01 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 86, Issue 57 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: Receiving musinc on hold instead of ring (Alejandro Recarey) 2. Re: Receiving musinc on hold instead of ring (Tarek Sawah) 3. Limit outbond calls duration to 1 minute (salaheddine elharit) 4. FreeTDS and MS-SQL with Asterisk RealTime (Reuben Fine) 5. Re: Limit outbond calls duration to 1 minute (Paul Belanger) 6. Re: Limit outbond calls duration to 1 minute (Tarek Sawah) 7. Re: Limit outbond calls duration to 1 minute (salaheddine elharit) 8. Re: Limit outbond calls duration to 1 minute (salaheddine elharit) 9. Re: Limit outbond calls duration to 1 minute (Danny Nicholas) 10. Re: Limit outbond calls duration to 1 minute (Tarek Sawah) 11. Re: Limit outbond calls duration to 1 minute (salaheddine elharit) 12. Re: Limit outbond calls duration to 1 minute (Tarek Sawah) 13. res_ODBC and failover (Bryant Zimmerman) 14. Anybody using BinFone Telecom? (ft...@mindspring.com) 15. Increasing the fxorxgain and fxotxgain for the hardware of the digium card (bilal ghayyad) ---------------------------------------------------------------------- Message: 1 Date: Wed, 28 Sep 2011 18:59:39 +0200 From: Alejandro Recarey <alexreca...@gmail.com> Subject: Re: [asterisk-users] Receiving musinc on hold instead of ring To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <ca+zbgr8ars-e2m+wvmm4bvgwuknbuovdvamgbc0qr4xm2q8...@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 > this is related to your carrier's SIP messages as they are sending a > sendonly attribute instead of sendrecv (taking a wild guess here) your > asterisk will act as if the call was placed on hold thus the MOH butts in. > an sip debug log for a similar call will be more helpful? Thanks for the answer Tarek! I will try to obtain a full SIP trace tonight. If the problem is indeed that the carrier is sending the sendonly attribute in the SDP instead of sendrecv, what can I do? Is there anything I can configure on my side? Thanks again, Alex ------------------------------ Message: 2 Date: Wed, 28 Sep 2011 17:34:11 +0000 From: Tarek Sawah <tareksa...@hotmail.com> Subject: Re: [asterisk-users] Receiving musinc on hold instead of ring To: Asterisk Users <asterisk-users@lists.digium.com> Message-ID: <snt121-w34e19228122b3a2aca6e1dca...@phx.gbl> Content-Type: text/plain; charset="iso-8859-1" i have faced this problem with one of the major VoIP whole providers in India .. they have a new platform with Sonus switches.. which does not support sendrecv media attribute .. however a work around that may work for you .. is enabling re-invite on their peer. let me know if this works out for you. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 > From: alexreca...@gmail.com > Date: Wed, 28 Sep 2011 18:59:39 +0200 > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Receiving musinc on hold instead of ring > > > this is related to your carrier's SIP messages as they are sending a > > sendonly attribute instead of sendrecv (taking a wild guess here) > > your asterisk will act as if the call was placed on hold thus the MOH butts > > in. > > an sip debug log for a similar call will be more helpful? > > Thanks for the answer Tarek! I will try to obtain a full SIP trace > tonight. If the problem is indeed that the carrier is sending the > sendonly attribute in the SDP instead of sendrecv, what can I do? Is > there anything I can configure on my side? > > Thanks again, > > Alex > > -- > _____________________________________________________________________ > -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/ce5699d8/attachment-0001.htm> ------------------------------ Message: 3 Date: Wed, 28 Sep 2011 17:59:27 +0000 From: salaheddine elharit <salah.elharit...@gmail.com> Subject: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <cahexamvgcv8fev2pljeigqceacohf7gnydybq07tn3a_z9g...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => 222,n,AbsoluteTimeout(60) exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 222,n,Dial(SIP/${EXTEN},,KkTt) exten => 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/07526f4d/attachment-0001.htm> ------------------------------ Message: 4 Date: Wed, 28 Sep 2011 18:01:35 +0000 From: Reuben Fine <f...@allmysons.com> Subject: [asterisk-users] FreeTDS and MS-SQL with Asterisk RealTime To: "asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com> Message-ID: <8bb5d54201ff7a45ab99a14d0c37c166077...@ch1prd0702mb107.namprd07.prod.outlook.com> Content-Type: text/plain; charset="us-ascii" We have successfully setup and tested integration between Asterisk and MS-SQL. We are currently running about 70 simultaneous calls throughout the day however after some time our MS-SQL server (Windows 2008 64bit, SQL Server 2008) starts to increase it's memory usage exponentially. The MS-SQL server CPU also pegs at around 90%+ and becomes unresponsive and cannot accept new connections. We are running Asterisk 1.8.6 currently. FreeTDS version is 4.2 and UnixODBC is 2.2.12. The kernel information is : Linux 2.6.27.25-78.2.56.fc9.i686.PAE #1 SMP Thu Jun 18 12:36:07 EDT 2009 i686 i686 i386 GNU/Linux. We are using Realtime and using FreeTDS to connect to the MS-SQL server where we control sip users, voicemail and so forth. This works fine however when we enable CEL and CDR into MS-SQL the server begins to grow in memory usage / CPU usage until the SQL server halts and stops taking new requests. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/fa04db5e/attachment-0001.htm> ------------------------------ Message: 5 Date: Wed, 28 Sep 2011 14:02:04 -0400 From: Paul Belanger <pabelan...@digium.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <4e83611c.1010...@digium.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 11-09-28 01:59 PM, salaheddine elharit wrote: > hello list > > > i have configured a sip account in order to do an outbound calls and i > want to force a hang up after 1 min for 222 sip > > > in extensions.conf i have > > > exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) > > exten => 222,n,AbsoluteTimeout(60) > > > exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) > > exten => 222,n,Dial(SIP/${EXTEN},,KkTt) > > exten => 222,n,Hangup(); > > could you please see this code and tell me waht is wrong > *CLI> core show application Dial Look at the 'L' flag -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org ------------------------------ Message: 6 Date: Wed, 28 Sep 2011 18:08:59 +0000 From: Tarek Sawah <tareksa...@hotmail.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users <asterisk-users@lists.digium.com> Message-ID: <snt121-w40b9d1e407cfd99f3a2158ca...@phx.gbl> Content-Type: text/plain; charset="iso-8859-1" have a look at the following: "L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional." source http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 28 Sep 2011 17:59:27 +0000 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit outbond calls duration to 1 minute hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => 222,n,AbsoluteTimeout(60) exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 222,n,Dial(SIP/${EXTEN},,KkTt) exten => 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/2f6340e0/attachment-0001.htm> ------------------------------ Message: 7 Date: Wed, 28 Sep 2011 18:09:32 +0000 From: salaheddine elharit <salah.elharit...@gmail.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <cahexamu+cno_phduhxixstib1oyefwy-mfjfgyfoldy6yhx...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" i have this when L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are left. Repeat the warning every 'z' ms. The following special variables can be used with this option: * LIMIT_PLAYAUDIO_CALLER yes|no (default yes) Play sounds to the caller. * LIMIT_PLAYAUDIO_CALLEE yes|no Play sounds to the callee. * LIMIT_TIMEOUT_FILE File to play when time is up. * LIMIT_CONNECT_FILE File to play when call begins. * LIMIT_WARNING_FILE File to play as warning if 'y' is defined. The default is to say the time remaining. but i don't understand what i can do to solve this thanks 2011/9/28 Paul Belanger <pabelan...@digium.com> > On 11-09-28 01:59 PM, salaheddine elharit wrote: > >> hello list >> >> >> i have configured a sip account in order to do an outbound calls and i >> want >> to force a hang up after 1 min for 222 sip >> >> >> in extensions.conf i have >> >> >> exten => 222,1,MixMonitor(sip_${EXTEN}_**${UNIQUEID}.wav|av(0}V(0)) >> >> exten => 222,n,AbsoluteTimeout(60) >> >> >> exten => 222,n,Set(AUDIOHOOK_INHERIT(**MixMonitor)=yes) >> >> exten => 222,n,Dial(SIP/${EXTEN},,KkTt) >> >> exten => 222,n,Hangup(); >> >> could you please see this code and tell me waht is wrong >> >> *CLI> core show application Dial > > Look at the 'L' flag > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > ______________________________**______________________________**_________ > -- 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/8cece08a/attachment-0001.htm> ------------------------------ Message: 8 Date: Wed, 28 Sep 2011 18:22:57 +0000 From: salaheddine elharit <salah.elharit...@gmail.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <CAHexamuXv0Awg1nDP+4-8t4X6=Zf9qEvGH3aN=y8xemgjwq...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" but there is no exemple for when i must put X in order to limit the call can you please give me an exemple regards 2011/9/28 Tarek Sawah <tareksa...@hotmail.com> > have a look at the following: > "*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, > repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional." > > > source > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial > > Tarek Sawah > > Information Technology Adviser > > Integrated Digital Systems > > CCNP, MCSE, RHCE, TELECOM > > USA: +1 386 492 9993 > > > > ------------------------------ > Date: Wed, 28 Sep 2011 17:59:27 +0000 > From: salah.elharit...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Limit outbond calls duration to 1 minute > > > hello list > > > i have configured a sip account in order to do an outbound calls and i want > to force a hang up after 1 min for 222 sip > > > in extensions.conf i have > > exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) > exten => 222,n,AbsoluteTimeout(60) > > exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) > exten => 222,n,Dial(SIP/${EXTEN},,KkTt) > exten => 222,n,Hangup(); > could you please see this code and tell me waht is wrong > thanks and regards > > > > -- _____________________________________________________________________ -- > Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE > or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/17490dc7/attachment-0001.htm> ------------------------------ Message: 9 Date: Wed, 28 Sep 2011 13:25:14 -0500 From: "Danny Nicholas" <da...@debsinc.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <013d01cc7e0b$f75ae110$e610a330$@debsinc.com> Content-Type: text/plain; charset="us-ascii" As I read this, the following should be correct: exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(60000)) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Wednesday, September 28, 2011 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute but there is no exemple for when i must put X in order to limit the call can you please give me an exemple regards 2011/9/28 Tarek Sawah <tareksa...@hotmail.com> have a look at the following: "L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional." source http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 _____ Date: Wed, 28 Sep 2011 17:59:27 +0000 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit outbond calls duration to 1 minute hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => 222,n,AbsoluteTimeout(60) exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 222,n,Dial(SIP/${EXTEN},,KkTt) exten => 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _____________________________________________________________________ -- 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 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 <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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/8110a2c6/attachment-0001.htm> ------------------------------ Message: 10 Date: Wed, 28 Sep 2011 18:31:50 +0000 From: Tarek Sawah <tareksa...@hotmail.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users <asterisk-users@lists.digium.com> Message-ID: <snt121-w2278c3a876143d267f4929ca...@phx.gbl> Content-Type: text/plain; charset="iso-8859-1" exten => 222,n,Dial(SIP/${EXTEN},,KkTtLL(60000:30000:10000)) this will call the extension and sets the limit to 60000MS which equals 60 seconds.. and will inform the caller of his remaining time when he has only 30 seconds left.. and will repeat the notification every ten seconds (this is an over do and playing such sounds files at this rate will consume the resources!) Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 28 Sep 2011 18:22:57 +0000 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute but there is no exemple for when i must put X in order to limit the call can you please give me an exemple regards 2011/9/28 Tarek Sawah <tareksa...@hotmail.com> have a look at the following: "L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional." source http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 28 Sep 2011 17:59:27 +0000 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit outbond calls duration to 1 minute hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => 222,n,AbsoluteTimeout(60) exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 222,n,Dial(SIP/${EXTEN},,KkTt) exten => 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/6280e8b0/attachment-0001.htm> ------------------------------ Message: 11 Date: Wed, 28 Sep 2011 18:32:28 +0000 From: salaheddine elharit <salah.elharit...@gmail.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <cahexamudxvkyztrp4jc4r1o+ubn7kvlgu46vy77m4xv5r3z...@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" sorry but the issue still the same there is no hangup after 1Min regards 2011/9/28 Danny Nicholas <da...@debsinc.com> > As I read this, the following should be correct:**** > > exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(60000)) > > **** > > ** ** > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine > elharit > *Sent:* Wednesday, September 28, 2011 1:23 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Limit outbond calls duration to 1 minute** > ** > > ** ** > > but there is no exemple for when i must put X in order to limit the call** > ** > > **** > > can you please give me an exemple**** > > **** > > regards**** > > 2011/9/28 Tarek Sawah <tareksa...@hotmail.com>**** > > have a look at the following: > "*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, > repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional." > > > source > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial > > Tarek Sawah > > Information Technology Adviser > > Integrated Digital Systems > > CCNP, MCSE, RHCE, TELECOM > > USA: +1 386 492 9993 > > > **** > ------------------------------ > > Date: Wed, 28 Sep 2011 17:59:27 +0000 > From: salah.elharit...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Limit outbond calls duration to 1 minute **** > > ** ** > > hello list **** > > **** > > i have configured a sip account in order to do an outbound calls and i want > to force a hang up after 1 min for 222 sip**** > > **** > > **** > > in extensions.conf i have **** > > **** > > exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) > exten => 222,n,AbsoluteTimeout(60) > > exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) > exten => 222,n,Dial(SIP/${EXTEN},,KkTt) > exten => 222,n,Hangup(); > could you please see this code and tell me waht is wrong > thanks and regards**** > > **** > > **** > > ** ** > > -- _____________________________________________________________________ -- > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/958e501c/attachment-0001.htm> ------------------------------ Message: 12 Date: Wed, 28 Sep 2011 18:37:15 +0000 From: Tarek Sawah <tareksa...@hotmail.com> Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute To: Asterisk Users <asterisk-users@lists.digium.com> Message-ID: <snt121-w5177d57efc4ccc2c0b51dca...@phx.gbl> Content-Type: text/plain; charset="iso-8859-1" one adjustment i would suggest is using (|) instead of (,) exten => 222,n,Dial(SIP/${EXTEN}||KkTtL(60000)) Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 28 Sep 2011 18:32:28 +0000 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute sorry but the issue still the same there is no hangup after 1Min regards 2011/9/28 Danny Nicholas <da...@debsinc.com> As I read this, the following should be correct: exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(60000)) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Wednesday, September 28, 2011 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute but there is no exemple for when i must put X in order to limit the call can you please give me an exemple regards 2011/9/28 Tarek Sawah <tareksa...@hotmail.com> have a look at the following: "L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional." source http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Wed, 28 Sep 2011 17:59:27 +0000 From: salah.elharit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Limit outbond calls duration to 1 minute hello list i have configured a sip account in order to do an outbound calls and i want to force a hang up after 1 min for 222 sip in extensions.conf i have exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => 222,n,AbsoluteTimeout(60) exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 222,n,Dial(SIP/${EXTEN},,KkTt) exten => 222,n,Hangup(); could you please see this code and tell me waht is wrong thanks and regards -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/00cb004d/attachment-0001.htm> ------------------------------ Message: 13 Date: Wed, 28 Sep 2011 16:30:37 -0400 From: "Bryant Zimmerman" <brya...@zktech.com> Subject: [asterisk-users] res_ODBC and failover To: <asterisk-users@lists.digium.com> Message-ID: <3689ea1e$37cd9d4c$1d6777e$@com> Content-Type: text/plain; charset="us-ascii" I am toying with res_ODBC. currently I am using dns=ODBCvalue. Is there a way to fail this over to another dns value in the event the a primary is off line. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/03a56998/attachment-0001.htm> ------------------------------ Message: 14 Date: Wed, 28 Sep 2011 18:33:49 -0400 From: "ft...@mindspring.com" <ft...@mindspring.com> Subject: [asterisk-users] Anybody using BinFone Telecom? To: asterisk-users@lists.digium.com Message-ID: <4e83a0cd.7070...@mindspring.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Does anyone have any experience with BinFone for IAX termination? They good look on the website, but I'm looking for any comments. ------------------------------ Message: 15 Date: Wed, 28 Sep 2011 15:59:31 -0700 (PDT) From: bilal ghayyad <bilmar...@yahoo.com> Subject: [asterisk-users] Increasing the fxorxgain and fxotxgain for the hardware of the digium card To: asterisk-users@lists.digium.com Message-ID: <1317250771.98667.yahoomailclas...@web162014.mail.bf1.yahoo.com> Content-Type: text/plain; charset=us-ascii Hi All; In the zaptel, we were increasing the gain of the voice volume at the hardware level from the /etc/zaptel and /etc/modprob.conf files, but now we are using DAHDI, so where to do the same thing? I am looking actually to increase the volume at hardware level and not software to avoid the DTMF detection problem and to have better voice quality. Any advise? Regards Bilal ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 86, Issue 57 ********************************************** -- _____________________________________________________________________ -- 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
