Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Mark Murawski
On 12/27/2012 07:36 PM, Ron Wheeler wrote: On 27/12/2012 3:14 PM, Carlos Alvarez wrote: On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar mailto:maill...@lightspeed.ca>> wrote: This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obviou

Re: [asterisk-users] Paging for Praying

2012-12-27 Thread Steve Edwards
Please don't top-post. On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad wrote: How can I have Paging on Asterisk to call for pray? The pray is 5 times in a day and there is a timing for pray (actually it can be existed in a text file or database for the next 2 or 5 years). [mailto:asterisk-

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Ron Wheeler
On 27/12/2012 3:14 PM, Carlos Alvarez wrote: On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar > wrote: This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our proc

Re: [asterisk-users] Paging for Praying

2012-12-27 Thread bilal ghayyad
Thanks a lot for your kindly reply and help. Really I did not understand why you need to place them in the /var/spool/asterisk/outgoing? Regards Bilal --- > > I would set up 5 shell files called pray1.sh, pray2.sh, etc > and then set up > 5 entries in /etc/crontab to run them at t

[asterisk-users] Cisco AS5300 - no incoming sound

2012-12-27 Thread Mickael MONSIEUR
Hello, I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem. Sound from POTS -> Asterisk does not work. (In the sense Asterisk -> POTS it works!!) The problem lies in two directions (call initiated from the Asterisk or POTS) I have no firewall between Asterisk and Cisco. (it's a

Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through

2012-12-27 Thread Bryant Zimmerman
We process a lot of t.38 with Level 3 and our Asterisk servers. We do not prefer Adtran end points for t.38. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Danny Nicholas" Sent: Thursday, December 27, 2012 4:51 PM To: "Asterisk

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Larry Moore
On 28/12/2012 5:45 AM, Eric Wieling wrote: udptl.conf settings: [general] udptlstart=4000 udptlend=4998 udptlchecksums=no udptlfecentries = 3 udptlfecspan = 3 use_even_ports = yes T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 sip.conf settings: directmedia=yes faxdetect = no t38pt_ud

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Leandro Dardini
I usually set directmedia=yes with good results... Leandro 2012/12/27 Christopher Harrington > On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling wrote: > >> sip.conf settings: >> directmedia=yes >> >> > I know you've said you tried it both ways, but consensus seems to be that > directmedia needs t

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Christopher Harrington
On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling wrote: > sip.conf settings: > directmedia=yes > > I know you've said you tried it both ways, but consensus seems to be that directmedia needs to be =no when using UDPTL. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 --

Re: [asterisk-users] Call Forwarding / Follow-Me on PRI

2012-12-27 Thread Carlos Alvarez
On Thu, Dec 27, 2012 at 2:41 PM, Barry D. Hassler wrote: > Friends, > > Curious if others have run into this scenario, and can shed further light > on it. I am working with an installed base of systems using PRI circuits > from several carriers, and the symptoms I relate occur across the board. >

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Eric Wieling
udptl.conf settings: [general] udptlstart=4000 udptlend=4998 udptlchecksums=no udptlfecentries = 3 udptlfecspan = 3 use_even_ports = yes T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 sip.conf settings: directmedia=yes faxdetect = no t38pt_udptl=yes,redundancy,maxdatagram=400 -O

[asterisk-users] Call Forwarding / Follow-Me on PRI

2012-12-27 Thread Barry D. Hassler
Friends, Curious if others have run into this scenario, and can shed further light on it. I am working with an installed base of systems using PRI circuits from several carriers, and the symptoms I relate occur across the board. Most carriers are restricting CALLING Number ID to be one of the num

Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through

2012-12-27 Thread Danny Nicholas
Not certain that you actually do. I do know that T.38 can be a "dental" experience with Asterisk, but some folks have succeeded with it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursd

Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through

2012-12-27 Thread Eric Wieling
I was not aware you needed SpanDSP for T.38 passthrough.. How will that work with the UDPTL packets not going through Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday,

Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through

2012-12-27 Thread Danny Nicholas
#1 I assume you have spandsp installed #2 I'm guessing you got some hints from this thread - https://issues.asterisk.org/jira/browse/ASTERISK-18394 ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Larry Moore
On 28/12/2012 4:59 AM, Larry Moore wrote: On 28/12/2012 1:55 AM, Eric Wieling wrote: . . directmedia=no t38pt_udptl=no Hmm, the t38pt_udptl will need to be set to yes, this was set to no for non T.38 capable devices I had set faxdetect=no in the peer's configuration for the T.38 capabl

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Larry Moore
On 28/12/2012 1:55 AM, Eric Wieling wrote: We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involve

Re: [asterisk-users] CHANNEL(t38passthrough) is 0

2012-12-27 Thread Steve Totaro
Do you have reinvite allowed? That was an issue on one of my installations if I am remembering correctly. Any debug, logs, confs that would help? Thanks, Steve Totaro On Thu, Dec 27, 2012 at 12:15 PM, Eric Wieling wrote: > Setting directmedia=no does not help. The calls still go through and t

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Carlos Alvarez
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar wrote: > This past holiday weekend has resulted in some real groaners when it comes > to bugs in our dialplan, making obvious the need for some changes in our > procedures. > > First, our hours of operation for Christmas Eve, Christmas, Boxing Day and

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Danny Nicholas
I would say that the database method has the advantage over GotoIfTime in that it should stay the same between releases. More headache on the front end, but easier once it is up and running. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@li

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Mitch Claborn
We bypass this problem by storing business hours and holidays in a database table. We use an ODBC call to return whether or not to play the "day" or "night" greeting based on the database. We also store the name of a custom greeting file to play. The database is fairly easy to manipulate wit

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Logan Bibby
I'm a fan of your method. I haven't had good luck with GotoIfTime in the past. A lot of my dialplan is actually handled by an AGI script. I've always found that to be the easiest. On Thu, Dec 27, 2012 at 2:00 PM, Doug Lytle wrote: > Ernie Dunbar wrote: > >> This appears to boil down to "alway

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Danny Nicholas
The "simplest" way to address this kind of change is to test it a week (month) or so in advance on your test machine (we have VM's set up to mock our live machines). A protection against "last minute" changes is to have this kind of thing controlled by variables so you can possibly even avoid dial

Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Doug Lytle
Ernie Dunbar wrote: This appears to boil down to "always remember to test it at the time that it becomes relevant". But if I was the kind of person who always remembered to do things at the right time, then there would never be a need for computers to do jobs like this in the first place. I

[asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?

2012-12-27 Thread Ernie Dunbar
This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay,

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Eric Wieling
We are using t38pt_udptl=yes,redundancy,maxdatagram=400 Without the maxdatagram we get errors in the CLI. We also tried using FEC instead of redundancy, but no difference. -Original Message- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 2:23

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Christopher Harrington
Last thing to check, just for sanity's sake: t38pt_udptl=yes in sip.conf? It defaults to off. On Thu, Dec 27, 2012 at 12:32 PM, Eric Wieling wrote: > It does not appear to make any difference. Calls are still failing. > > -Original Message- > From: Christopher Harrington [mailto:ch.

Re: [asterisk-users] Vxml record voice parameter

2012-12-27 Thread Christopher Harrington
If VoiceGlue is the software making the recording, then that's where you need to look for support. Try https://github.com/voiceglue/voiceglue/issues and http://www.voiceglue.org/mailing-list/ . On Thu, Dec 27, 2012 at 12:27 PM, ulvi cesur wrote: > I am using voiceglue to record voice. > > VXML

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Eric Wieling
It does not appear to make any difference. Calls are still failing. -Original Message- From: Christopher Harrington [mailto:ch...@acsdi.com] Sent: Thursday, December 27, 2012 1:20 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users

Re: [asterisk-users] Vxml record voice parameter

2012-12-27 Thread ulvi cesur
I am using voiceglue to record voice. VXML : I can take rec parameter but it is not file. rec is audio/basic:len(123123):p0x5a6e6241. 2012/12/26 Christopher Harrington > On Tue, Dec 25, 2012 at 8:57 AM, ulvi cesur wrote: > >> Hi, I am working on vxml to record voice. I have trouble wit

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Christopher Harrington
True, but it should bypass Asterisk when possible for SIP streams and may solve your problem. On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling wrote: > We have directrtpsetup=no because the comments in the sample config > indicates it does not work in all situations. > > -Original Message

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Eric Wieling
We have directrtpsetup=no because the comments in the sample config indicates it does not work in all situations. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher Harrington Sent: Thursday, Decembe

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Christopher Harrington
directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling wrote: > We have set directmedia=yes as well as directmedia=no. There is no NAT > involved. > > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Eric Wieling
We have set directmedia=yes as well as directmedia=no. There is no NAT involved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users

Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Leandro Dardini
Have you configured the canreinvite=yes in sip peer? I am currently off work for two days, but a 100% fail means a configuration problem for sure. Leandro 2012/12/27 Eric Wieling > We are offering $100 (paid via paypal or check) to the first person who > assists us in successfully sending and

[asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

2012-12-27 Thread Eric Wieling
We are offering $100 (paid via paypal or check) to the first person who assists us in successfully sending and receiving faxes in the setup described below. Offer expires Dec 31. We are a direct customer of Level 3, there is no other carrier involved. What we want to work: Level 3 T.38 T

Re: [asterisk-users] stop log/debug messages into /var/log/messages

2012-12-27 Thread Steve Edwards
On Thu, 27 Dec 2012, [Digital^Dude] ® wrote: I disabled all logger channels but still it logs to /var/log/messages. Any hints? What version of Asterisk? What does 'logger show channels' show? Any chance your startup script pipes output through the logger shell command? -- Thanks in advanc

Re: [asterisk-users] dahdi timing source multiple cards

2012-12-27 Thread Dave George
Thanks Matt. The suggestion helped. No more slip erros. Dave > Original Message > Subject: Re: [asterisk-users] dahdi timing source multiple cards > From: Matthew Fredrickson > Date: Fri, December 21, 2012 3:41 pm > To: asterisk-users@lists.digium.com > > > You must make s

Re: [asterisk-users] CHANNEL(t38passthrough) is 0

2012-12-27 Thread Eric Wieling
Setting directmedia=no does not help. The calls still go through and the fax still fails after switching to T.38. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, December 27, 20

Re: [asterisk-users] Paging for Praying

2012-12-27 Thread Danny Nicholas
I would set up 5 shell files called pray1.sh, pray2.sh, etc and then set up 5 entries in /etc/crontab to run them at the specified time daily. The file pray1.sh should look something like this: #!/bin/sh cp /pray1/*.call /tmp mv /tmp/*.call /var/spool/asterisk/outgoing the entry in /etc/cro

Re: [asterisk-users] stop log/debug messages into /var/log/messages

2012-12-27 Thread Danny Nicholas
If it is writing to /v/l/m, then it is coming from somewhere else. All Asterisk messages go to /v/l/asterisk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, December 27, 2012 3:32 AM To: Asterisk User

Re: [asterisk-users] Asterisk with Cisco 887M

2012-12-27 Thread Danny Nicholas
Shouldn't be difficult. You're just setting up the Cisco box as a SIP gateway. Here's a link to get you started. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b06gtw ay.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.c

Re: [asterisk-users] Asterisk with Cisco 887M

2012-12-27 Thread Carlos Alvarez
On Thu, Dec 27, 2012 at 9:10 AM, Edwin Quijada wrote: > Hi! > I am installing asterisk with my ISP but he give me a Cisco 887M router to > use for SIP conection. My problem is that I dont know how to link Asterisk > with this device because I dont have user/pass to use. > > Anybody has a cluee to

[asterisk-users] Asterisk with Cisco 887M

2012-12-27 Thread Edwin Quijada
Hi! I am installing asterisk with my ISP but he give me a Cisco 887M router to use for SIP conection. My problem is that I dont know how to link Asterisk with this device because I dont have user/pass to use. Anybody has a cluee to use CISCO 887M with Asterisk ? Thks!

[asterisk-users] CHANNEL(t38passthrough) is 0

2012-12-27 Thread Eric Wieling
I'm trying to get T.38 passthrough working on Asterisk 1.8.18.1. It isn't working. Calls go through and are answered, but the fax machines are unable to communicate. I checked the value of CHANNEL(t38passthrough) and it seems to always be 0. One side is Level 3 T.38 TN and the other side

Re: [asterisk-users] stop log/debug messages into /var/log/messages

2012-12-27 Thread DHAVAL INDRODIYA
please refer logger.conf under /etc/asterisk and stop messages log for full. On Thu, Dec 27, 2012 at 2:43 PM, [Digital^Dude] ® wrote: > I disabled all logger channels but still it logs to /var/log/messages. > Any hints? > > -- >

[asterisk-users] stop log/debug messages into /var/log/messages

2012-12-27 Thread [Digital^Dude] ®
I disabled all logger channels but still it logs to /var/log/messages. Any hints? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Paging for Praying

2012-12-27 Thread Bharat Lalcheta
I dont think this is existed. However, its easy to build a script in php or perl or any other language which check time from file or database and generate call file which execute paging in asterisk. Just put this script in cron. Thats it... Regards, Bharat Lalcheta On Thu, Dec 27, 2012 at 1: