Re: [asterisk-users] Paging for Praying
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 1:29 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; 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). My question is compound from two parts: How can I have Automatic Page? The automatic page should happens by reading the time and check if the time is same as this time, then do the Page. How? Is it by cron? Someone told me that do a cron that call a script which will check the time, if the time came to do th Page, then do a Page. But really I do not know how this can be done and I do not know if this is already existed? Regards Bilal -- _ -- 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 -- Bharat Lalcheta -- _ -- 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] stop log/debug messages into /var/log/messages
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: 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] stop log/debug messages into /var/log/messages
please refer logger.conf under /etc/asterisk and stop messages log for full. On Thu, Dec 27, 2012 at 2:43 PM, [Digital^Dude] ® millennium@gmail.comwrote: 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: 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
[asterisk-users] CHANNEL(t38passthrough) is 0
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 is an Adtran NetVanta with POTS ports. What would cause Asterisk to not offer t38 passthrough on a channel? -- _ -- 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] Asterisk with Cisco 887M
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! -- _ -- 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] Asterisk with Cisco 887M
On Thu, Dec 27, 2012 at 9:10 AM, Edwin Quijada listas_quij...@hotmail.comwrote: 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 ? What are the symptoms of the problem? One way audio? No audio? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Asterisk with Cisco 887M
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.com] On Behalf Of Edwin Quijada Sent: Thursday, December 27, 2012 10:10 AM To: Asterisk Asterisk Subject: [asterisk-users] Asterisk with Cisco 887M 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! -- _ -- 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] stop log/debug messages into /var/log/messages
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 Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages please refer logger.conf under /etc/asterisk and stop messages log for full. On Thu, Dec 27, 2012 at 2:43 PM, [Digital^Dude] R millennium@gmail.com wrote: 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: 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] Paging for Praying
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/crontab would look like this 0 8 *** root /usr/bin/pray1.sh This would run pray1.sh at 8 am daily. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat Lalcheta Sent: Thursday, December 27, 2012 2:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging for Praying 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 1:29 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; 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). My question is compound from two parts: How can I have Automatic Page? The automatic page should happens by reading the time and check if the time is same as this time, then do the Page. How? Is it by cron? Someone told me that do a cron that call a script which will check the time, if the time came to do th Page, then do a Page. But really I do not know how this can be done and I do not know if this is already existed? Regards Bilal -- _ -- 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 -- Bharat Lalcheta -- _ -- 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] CHANNEL(t38passthrough) is 0
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, 2012 10:49 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CHANNEL(t38passthrough) is 0 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 is an Adtran NetVanta with POTS ports. What would cause Asterisk to not offer t38 passthrough on a channel? -- _ -- 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] dahdi timing source multiple cards
Thanks Matt. The suggestion helped. No more slip erros. Dave Original Message Subject: Re: [asterisk-users] dahdi timing source multiple cards From: Matthew Fredrickson cres...@digium.com Date: Fri, December 21, 2012 3:41 pm To: asterisk-users@lists.digium.com You must make sure that for each card, the timing parameter does not exceed the number of spans on the card (unless you're using a timing cable between cards). So you probably don't want to have anything above a 4 for the timing parameter... I see below that you have 5-12 listed in the timing parameter for the spans on the other cards. You probably want something more like this: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs span=5,1,0,esf,b8zs span=6,2,0,esf,b8zs span=7,3,0,esf,b8zs span=8,4,0,esf,b8zs span=9,1,0,esf,b8zs span=10,2,0,esf,b8zs span=11,3,0,esf,b8zs span=12,4,0,esf,b8zs Hope that helps. Matthew Fredrickson Digium, Inc. On 12/20/12 10:42 PM, Dave George wrote: I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72 span=4,4,0,esf,b8zs bchan=73-96 span=5,5,0,esf,b8zs bchan=97-120 span=6,6,0,esf,b8zs bchan=121-144 span=7,7,0,esf,b8zs bchan=145-168 span=8,8,0,esf,b8zs bchan=169-192 span=9,9,0,esf,b8zs bchan=193-216 span=10,10,0,esf,b8zs bchan=217-240 span=11,11,0,esf,b8zs bchan=241-264 span=12,12,0,esf,b8zs bchan=265-288 dahdi_scan : [1] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 1 name=TE4/0/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=1 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=25 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [3] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 3 name=TE4/0/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=49 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [4] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 4 name=TE4/0/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=73 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [5] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 1 name=TE4/1/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=97 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [6] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 2 name=TE4/1/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=121 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [7] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 3 name=TE4/1/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=145 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [8] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 4 name=TE4/1/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=169 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1)
Re: [asterisk-users] stop log/debug messages into /var/log/messages
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 advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com 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.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com 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.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 ewiel...@nyigc.com wrote: 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, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com 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.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Vxml record voice parameter
I am using voiceglue to record voice. VXML : record name=rec finalsilence=4s maxtime=30s beep=true dtmfterm=true filled submit enctype=*multipart*/*form*-*data* next=http://domain/getVxml/ namelist=rec method=post /filled /record I can take rec parameter but it is not file. rec is audio/basic:len(123123):p0x5a6e6241. 2012/12/26 Christopher Harrington ch...@acsdi.com On Tue, Dec 25, 2012 at 8:57 AM, ulvi cesur uce...@gmail.com wrote: Hi, I am working on vxml to record voice. I have trouble with getting url of recorded voice. I want to save and I am using java to get record parameter from url and it returns a string which is audio/basic:len(123123):p0x5a6e6241, but I want to get file object or base64 string with parameter or to relate returning string with path in asterisk server, are there any way to do this? How are you recording the audio in Asterisk? ChanSpy, Voicemail, etc? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com wrote: 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, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com 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.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Vxml record voice parameter
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 uce...@gmail.com wrote: I am using voiceglue to record voice. VXML : record name=rec finalsilence=4s maxtime=30s beep=true dtmfterm=true filled submit enctype=*multipart*/*form*-*data* next=http://domain/getVxml/ namelist=rec method=post /filled /record I can take rec parameter but it is not file. rec is audio/basic:len(123123):p0x5a6e6241. 2012/12/26 Christopher Harrington ch...@acsdi.com On Tue, Dec 25, 2012 at 8:57 AM, ulvi cesur uce...@gmail.com wrote: Hi, I am working on vxml to record voice. I have trouble with getting url of recorded voice. I want to save and I am using java to get record parameter from url and it returns a string which is audio/basic:len(123123):p0x5a6e6241, but I want to get file object or base64 string with parameter or to relate returning string with path in asterisk server, are there any way to do this? How are you recording the audio in Asterisk? ChanSpy, Voicemail, etc? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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 -- -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 ewiel...@nyigc.com wrote: 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com wrote: 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, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com 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.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 PM To: Eric Wieling Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com wrote: 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com wrote: 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, December 27, 2012 1:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through directrtpsetup=yes in sip.conf? On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling ewiel...@nyigc.com 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.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, December 27, 2012 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 ewiel...@nyigc.com 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 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. -- _ -- 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:
[asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?
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, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan was broken by me misspelling the correct dialplan to go to. Then our Boxing Day dialplan was broken when I copied and pasted the correct dialplan from one similar extension number to the other, like this: ; Christmas ; exten = 821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early,s,1) exten = 821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1) exten = 821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1) exten = 821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1) then failed to notice the problem until it was too late. Of course, that only applied on Boxing day and couldn't be noticed earlier, either. I suppose the first problem where I misspelt the dialplan can be solved by testing the dialplan in another extension and modifying the time to now + 2 minutes. But how can I avoid stupid errors in the extension number, when testing by definition requires that I change the extension number to and fro? 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. -- _ -- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
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 no longer use GotoIfTime for these events. I do database lookups based on date. At the beginning of each year, our HR department releases the holiday schedule and I enter them into the database. All inbound calls query the database to see if there is a match and jump to the appropriate sub-routine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
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
dialplan changes by controlling the variables with an AGI script.
In your case, the dialplan could have been written like this:
; Christmas
Exten = s,1,Set(christday=25)
Exten = s,n,Set(eveday=24)
Exten = s,n,Set(boxday=26)
Exten = s,n,Set(christmon=Dec)
Exten = s,n,set(christopen=9:30)
...
; exten =
821192,n,GotoIfTime(${christopen}-${christclose},*,${christday},${christ
mon}?ivr-lightspeed-tech-early,s,1)
exten =
821192,n,GotoIfTime(${eveopen}-${eveclose},*,${eveday},${christmon}?ivr-
lightspeed-day,s,1)
exten =
821192,n,GotoIfTime(*,*,${christday},${christmon}?ivr-lightspeed-after-h
ours,s,1)
exten =
821190,n,GotoIfTime(${boxopen}-${boxclose},*,${boxday},${christmon}?ivr-
lightspeed-day,s,1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Thursday, December 27, 2012 1:46 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do *you* test your changes to dialplans ruled
by GotoIfTime?
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, fine, whatever, I
fix.
Our Christmas Eve hours (made worse by being Monday this year) dialplan was
broken by me misspelling the correct dialplan to go to. Then our Boxing Day
dialplan was broken when I copied and pasted the correct dialplan from one
similar extension number to the other, like this:
; Christmas
; exten =
821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early,s,1)
exten =
821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1)
exten =
821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1)
exten =
821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1)
then failed to notice the problem until it was too late. Of course, that
only applied on Boxing day and couldn't be noticed earlier, either.
I suppose the first problem where I misspelt the dialplan can be solved by
testing the dialplan in another extension and modifying the time to now + 2
minutes. But how can I avoid stupid errors in the extension number, when
testing by definition requires that I change the extension number to and
fro?
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.
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Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?
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 supp...@drdos.info wrote: 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 no longer use GotoIfTime for these events. I do database lookups based on date. At the beginning of each year, our HR department releases the holiday schedule and I enter them into the database. All inbound calls query the database to see if there is a match and jump to the appropriate sub-routine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby, CEO Ke*o*bi Communications Tuscaloosa, Alabama -- _ -- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
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 with test data. Mitch On 12/27/2012 01:46 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 New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan was broken by me misspelling the correct dialplan to go to. Then our Boxing Day dialplan was broken when I copied and pasted the correct dialplan from one similar extension number to the other, like this: ; Christmas ; exten = 821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early,s,1) exten = 821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1) exten = 821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1) exten = 821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1) then failed to notice the problem until it was too late. Of course, that only applied on Boxing day and couldn't be noticed earlier, either. I suppose the first problem where I misspelt the dialplan can be solved by testing the dialplan in another extension and modifying the time to now + 2 minutes. But how can I avoid stupid errors in the extension number, when testing by definition requires that I change the extension number to and fro? 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. -- _ -- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
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...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Thursday, December 27, 2012 2:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime? 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 with test data. Mitch On 12/27/2012 01:46 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 New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan was broken by me misspelling the correct dialplan to go to. Then our Boxing Day dialplan was broken when I copied and pasted the correct dialplan from one similar extension number to the other, like this: ; Christmas ; exten = 821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early, s,1) exten = 821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1) exten = 821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1) exten = 821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1) then failed to notice the problem until it was too late. Of course, that only applied on Boxing day and couldn't be noticed earlier, either. I suppose the first problem where I misspelt the dialplan can be solved by testing the dialplan in another extension and modifying the time to now + 2 minutes. But how can I avoid stupid errors in the extension number, when testing by definition requires that I change the extension number to and fro? 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. -- _ -- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.cawrote: 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, fine, whatever, I fix. Boxing day??? Seriously? There's a holiday for people who beat each other up? TIL. But anyway the best way to test time-based rules is on a VM that has a copy of your configs, and just change the time. You can easily run a small VM to accomodate a copy of your main server on almost any computer. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] CHANNEL(t38passthrough) is 0
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 ewiel...@nyigc.com wrote: 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, 2012 10:49 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CHANNEL(t38passthrough) is 0 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 is an Adtran NetVanta with POTS ports. What would cause Asterisk to not offer t38 passthrough on a channel? -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. In udptl.conf try the following option ; ; Some VoIP providers will only accept an offer with an even-numbered ; UDPTL port. Set this option so that Asterisk will only attempt to use ; even-numbered ports when negotiating T.38. Default is no. use_even_ports = yes ; Looking at some old notes other options I set for some devices to be able to pass through T.38 in sip.conf were, directmedia=no t38pt_udptl=no May be worth checking the following; directrtpsetup=no Larry. -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
On 28/12/2012 4:59 AM, Larry Moore wrote: On 28/12/2012 1:55 AM, Eric Wieling wrote: . snip . directmedia=no t38pt_udptl=no snip 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 capable device, perhaps this was to prevent an attempt by Asterisk to redirect the call to the fax extension in the dialplan. Larry. -- _ -- 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] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through
#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 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] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through
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, December 27, 2012 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through #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 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] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through
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: Thursday, December 27, 2012 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through 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, December 27, 2012 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through #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 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
[asterisk-users] Call Forwarding / Follow-Me on PRI
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 numbers allocated to the associated circuit. This makes sense from a perspective of call-fraud prevention. We have clients that used call forwarding or follow-me extensively, and configured to send the ORIGINAL callerID as the Calling ID, so when the call shows up on their cell phone, it appears to be coming from the originator. This capability seems to be going away. Some legacy PRI carriers were not (and perhaps continue to be) so strict about it, and a recent client who was used to this feature, is now unhappy that their new PRI carrier does not allow any callingID other than one associated with the PRI. A workaround or RNIE (Redirecting Number Information Element) has been recommended as an alternative, but that does not appear to be standardized, nor implemented in asterisk. The only PBX vendors that appear to support this in even a limited sense are Cisco and Shoretel. I'm curious if others have encountered this same situation (I'm sure you have), or any other pertinent thoughts. Thanks in advance! -- Barry D. Hassler President, HCST http://www.hcst.com/ 937-427-9000 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Thursday, December 27, 2012 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through 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 involved. What we want to work: Level 3 T.38 TN - MSX/Nextone SBC - Asterisk 1.8.18.1 - Adtran NetVanta w/POTS and T.38 support. When we replace Asterisk with Kamailio faxes work fine. When we put Asterisk there instead, then faxes fail nearly 100% of the time. I see the switch to T.38 in the Adtran debug logs. We can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and sip.conf settings correct. In udptl.conf try the following option ; ; Some VoIP providers will only accept an offer with an even-numbered ; UDPTL port. Set this option so that Asterisk will only attempt to use ; even-numbered ports when negotiating T.38. Default is no. use_even_ports = yes ; Looking at some old notes other options I set for some devices to be able to pass through T.38 in sip.conf were, directmedia=no t38pt_udptl=no May be worth checking the following; directrtpsetup=no Larry. -- _ -- 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] Call Forwarding / Follow-Me on PRI
On Thu, Dec 27, 2012 at 2:41 PM, Barry D. Hassler barry.hass...@gmail.comwrote: 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. We have encountered it, and simply told the carriers to stop blocking it or lose the business. All but one did it, and we dropped their services. Don't know that there's a good work-around otherwise. Is there a reason you don't just go all SIP, where 98% of the service providers will accept any CLID? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com 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 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
I usually set directmedia=yes with good results... Leandro 2012/12/27 Christopher Harrington ch...@acsdi.com On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling ewiel...@nyigc.com 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 -- _ -- 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] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through
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_udptl=yes,redundancy,maxdatagram=400 From memory when I was doing this in March 2011 Asterisk would not allow a T.38 connection to successfully establish when canreinvite was set to yes, I did have NAT involved in my testing hence T.38 would be successful when canreinvite=no, the options to use now seeing as canreinvite is deprecated are; directmedia=no direcrtpsetup=no The T38Fax... options you have in udptl.conf are no longer supported. I have the T.38 Fax Gateway patch applied to my installation of 1.8.18.1 though I don't believe this will make any difference as I had got my T.38 relaying working prior to the patch. I have in my sip.conf; [general] t38pt_udptl=yes,redundancy,maxdatagram=1400 You may also want to enable; t38pt_usertpsource=yes Larry. -- _ -- 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] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through
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 da...@debsinc.com Sent: Thursday, December 27, 2012 4:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38Pass-Through 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: Thursday, December 27, 2012 3:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through 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, December 27, 2012 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] $100 Bounty: Level 3/Asterisk/Adtran T.38 Pass-Through #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 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
[asterisk-users] Cisco AS5300 - no incoming sound
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 LAN) Do you have any ideas? Thank you, Mickael -- _ -- 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] Paging for Praying
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 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/crontab would look like this 0 8 *** root /usr/bin/pray1.sh This would run pray1.sh at 8 am daily. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat Lalcheta Sent: Thursday, December 27, 2012 2:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging for Praying 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 1:29 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; 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). My question is compound from two parts: How can I have Automatic Page? The automatic page should happens by reading the time and check if the time is same as this time, then do the Page. How? Is it by cron? Someone told me that do a cron that call a script which will check the time, if the time came to do th Page, then do a Page. But really I do not know how this can be done and I do not know if this is already existed? Regards Bilal -- _ -- 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] How do *you* test your changes to dialplans ruled by GotoIfTime?
On 27/12/2012 3:14 PM, Carlos Alvarez wrote: On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar maill...@lightspeed.ca mailto:maill...@lightspeed.ca 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 New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Boxing day??? Seriously? There's a holiday for people who beat each other up? TIL. That is the day you box up all the crap you got and exchange it for what you really wanted. It is a religious holiday in the old British Commonwealth (probably Scottish in origin). Ron But anyway the best way to test time-based rules is on a VM that has a copy of your configs, and just change the time. You can easily run a small VM to accomodate a copy of your main server on almost any computer. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Paging for Praying
Please don't top-post. On Thu, Dec 27, 2012 at 1:29 PM, bilal ghayyad bilmar...@yahoo.com 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-users-boun...@lists.digium.com] On Behalf Of Bharat Lalcheta 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... From: Danny Nicholas da...@debsinc.com 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/crontab would look like this 0 8 *** root /usr/bin/pray1.sh This would run pray1.sh at 8 am daily. On Thu, 27 Dec 2012, bilal ghayyad wrote: 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? The appropriate solution needs a lot more detail to be useful. Is this just to remind you or is this the foundation of a new product for thousands of customers? Is there a message or verse associated with each of the 5 reminders or is 'time to pray' sufficient? Is there a penalty associated with missing a prayer like eternal damnation? (AMI is more robust than call files.) The answers would help guide you in deciding if a simple cron based shell script generating call files or a database driven AMI daemon is the best approach. In answer to your specific question, the call files need to be mv'ed into /v/s/a/o/ because: ) You need to use mv instead of cp because mv is an 'atomic' function* meaning it happens all at once so that Asterisk will not try to read an incomplete file. ) This is the default value of 'astspooldir.' You can specify a different location in asterisk.conf if needed. *) Assuming the source and destination are on the same filesystem. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?
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 maill...@lightspeed.ca mailto:maill...@lightspeed.ca 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 New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Boxing day??? Seriously? There's a holiday for people who beat each other up? TIL. That is the day you box up all the crap you got and exchange it for what you really wanted. It is a religious holiday in the old British Commonwealth (probably Scottish in origin). Ron But anyway the best way to test time-based rules is on a VM that has a copy of your configs, and just change the time. You can easily run a small VM to accomodate a copy of your main server on almost any computer. -- Carlos Alvarez TelEvolve 602-889-3003 What about using TESTTIME core show function TESTTIME -- _ -- 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
