[asterisk-users] ExternalIVR testing
I've opened a few bugs on ExternalIVR and added patches. The biggest issue is: https://issues.asterisk.org/view.php?id=16174 [patch] ExternalIVR does not handle arguments in a consistant manner Basically, this optimizes and fixes several different ways of calling ExternalIVR. If there is anyone who is using ExternalIVR today and/or is willing to test these patches I would appreciate the help. I will be glad to provide any help or explanation needed. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI call progress issue
I've got a couple of PRIs. When I call out on them from internal SIP phones, I will get ringing if the dialed number is ringing, but if the dialed number is busy I'll get dead air. Can anyone suggest ways to trouble shoot this? Don't seem to having any other problems with the PRIs. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI call progress issue
Apologies. Didn't mean to omit key information, I doubt it's a problem with * because everything else is working great so I was asking for help on troubleshooting the PRI. Anyway, here's the 411: Asterisk 1.4.20, CentOS 5.2 Service Providers: Quest Deltacom, Local Loops provided by Embarq What snippets of the config would be helpful? I've included zapata.conf and zaptel.conf below. zapata.conf: ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A102 port 1 [slot:8 bus:1 span:5] wanpipe5 switchtype=dms100 context=from-pstn group=0 signalling=pri_cpe channel =97-119 ;Sangoma A102 port 2 [slot:8 bus:1 span:6] wanpipe6 switchtype=dms100 context=from-pstn group=1 signalling=pri_cpe channel =121-143 zaptel.conf: # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:8 bus:1 span:5] wanpipe5 span=5,0,0,esf,b8zs bchan=97-119 hardhdlc=120 #Sangoma A102 port 2 [slot:8 bus:1 span:6] wanpipe6 span=6,0,0,esf,b8zs bchan=121-143 hardhdlc=144 Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, July 23, 2009 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI call progress issue David Ruggles wrote: is busy I'll get dead air. Can anyone suggest ways to trouble shoot this? Don't seem to having any other problems with the PRIs. It'd be nice to start with what version of Asterisk, what distro, who is your service provider and snippets of your config. On our PRIs we show busy when a line is busy. My systems are out of Michigan and Indiana. 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Duplicating existing PBX function
Right now, we have a pbx that auto-answers for extension-to-extension calls, but after the phone has been auto answered, lets the caller press one to cause the phone to start ringing. (for example, the person's not in their office so you want it to ring through to voicemail) I'm able to duplicate the auto answer using the SIP add header function since I have Grandstream phones. I assuming what I need to do is setup a feature that executes a macro when the user presses one. This macro would hangup the callee and then redial the callee without sending the extra SIP header so the phone rings instead of auto-answers. Any suggestions on how to do this? If there's another way, I'm open to that as well. TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gxp 2000 softkey question
I have a function *1 that starts and stops recording in a call. I use a function so I can use MixMonitor. It works well, however I would like to make it a little more integrated for my users. We have GXP 200 hardphones. So far I've been able to configure a softkey using the speeddial option to dial *1 during a call. I also have setup another key to monitor the status of recording (on/off) using the BLF function and the devstate function. (new in 1.6 back ported to 1.4) However I seem to be unable to combine these functions in to a single key. Can anyone offer any assistance? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone question
I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP vs RTP destination IP
Thx! That did it. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E. Johansson Sent: Friday, April 03, 2009 4:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP vs RTP destination IP 2 apr 2009 kl. 17.45 skrev David Ruggles: Is it possible to have asterisk override the connection information embedded in a SIP 200 packet with the registration information? I have multihomed machines with softphones and they register just fine and sip works fine, but the RTP packets get sent to the ip from the SIP connection information and the softphones are sending the wrong ip. I can't find an option in the softphone to change ip it sends. If you turn on NAT support, we will ignore all IP addresses in the 200 OK and just send our media directly to wherever the other end sends it from. /O --- * Olle E. Johansson - o...@edvina.net * Asterisk/OpenSER/Kamailio Training http://edvina.net/training/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP vs RTP destination IP
Is it possible to have asterisk override the connection information embedded in a SIP 200 packet with the registration information? I have multihomed machines with softphones and they register just fine and sip works fine, but the RTP packets get sent to the ip from the SIP connection information and the softphones are sending the wrong ip. I can't find an option in the softphone to change ip it sends. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird sip problem
There are no sip packets created to the hard phone (I'm using a softphone and those sip packets are there) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Sent: Saturday, March 28, 2009 9:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Weird sip problem What does the SIP debug say when you attempt to dial? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Saturday, March 28, 2009 4:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Weird sip problem *bump* and more information: I packet sniffed the server during the attempt to call the phone and * never sends a packet to the phone before generating the status is 'UNKNOWN' message. I assume this means that * somehow knows the phone is unavailable, which doesn't make sense to me sense sip show peers shows the phone as OK. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Friday, March 27, 2009 12:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Weird sip problem I've got a weird problem: I've added a new phone and sip show peers shows a status of OK (x ms) but when I dial it I get status is 'UNKNOWN' Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: 270.11.31/2028 - Release Date: 03/28/09 07:16:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird sip problem
If it makes a difference the phone is a gxp 2000 Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Monday, March 30, 2009 1:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Weird sip problem There are no sip packets created to the hard phone (I'm using a softphone and those sip packets are there) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Sent: Saturday, March 28, 2009 9:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Weird sip problem What does the SIP debug say when you attempt to dial? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Saturday, March 28, 2009 4:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Weird sip problem *bump* and more information: I packet sniffed the server during the attempt to call the phone and * never sends a packet to the phone before generating the status is 'UNKNOWN' message. I assume this means that * somehow knows the phone is unavailable, which doesn't make sense to me sense sip show peers shows the phone as OK. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Friday, March 27, 2009 12:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Weird sip problem I've got a weird problem: I've added a new phone and sip show peers shows a status of OK (x ms) but when I dial it I get status is 'UNKNOWN' Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: 270.11.31/2028 - Release Date: 03/28/09 07:16:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird sip problem
*bump* and more information: I packet sniffed the server during the attempt to call the phone and * never sends a packet to the phone before generating the status is 'UNKNOWN' message. I assume this means that * somehow knows the phone is unavailable, which doesn't make sense to me sense sip show peers shows the phone as OK. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Friday, March 27, 2009 12:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Weird sip problem I've got a weird problem: I've added a new phone and sip show peers shows a status of OK (x ms) but when I dial it I get status is 'UNKNOWN' Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird sip problem
I've got a weird problem: I've added a new phone and sip show peers shows a status of OK (x ms) but when I dial it I get status is 'UNKNOWN' Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Provisioning GXP 2000
I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? TIA!!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
I've worked with VoIP Supply several times in the past. I've been very pleased with their service. And if you compare the prices of the two phones you mention: Polycom IP 320 330 a difference of 109.94 vs 106 and 84.95 vs 83 seems to disperse the allegation of being overpriced. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Joakimsen Sent: Wednesday, March 18, 2009 3:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Good phone near $125 On Mon, Mar 16, 2009 at 20:26, Marc Charbonneau timebandit...@gmail.com wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply Or the Polycom 320 -- same phone as the 330, both have PoE support, 320 has 1 Ethernet port, 330 has two ports (built in switch) Now that I have to reply to your message, may I suggest telephonydepot.com. They have the 330 for $106 and the 320 for $83. FWIW VoIP supply are horrible (and overpriced.) They took 6 months to RMA a phone, and even then they didn't do what I requested. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
When I was first looking at Aastra, over a year ago, I thought there was some talk that Aastra was more supportive of asterisk then most vendors. Is this still true? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: Cory Andrews [mailto:c...@voipsupply.com] Sent: Monday, March 16, 2009 6:37 PM To: da...@safedatausa.com Subject: RE: [asterisk-users] Good phone near $125 David - not sure if you have any specific requirements in terms of # of lines or other features, but the Polycom IP330 and Linksys SPA942 are excellent phones which are in your price range. http://www.voipsupply.com/polycom-ip-330 http://www.voipsupply.com/linksys-spa942 Also the Grandstream GXP2010 and Aastra 6731i http://www.voipsupply.com/grandstream-gxp2010 http://www.voipsupply.com/catalog/product/view/id/7991/s/aastra-6731i-ip-pho ne/ Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Monday, March 16, 2009 6:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Good phone near $125 I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.13/1999 - Release Date: 03/16/09 07:04:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra 9133i programmable buttons (* 4.1.23)
Is it possible to control the light on a programmable button without the blf option? I'm using a programmable button to turn call recording on and off and I would like the light to indicate the status. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Good phone near $125
I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)
I'm sorry, but it looks like it's working correctly now. I will update the bug if I am able to verify any problems. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Friday, March 13, 2009 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4) David Ruggles wrote: The patch doesn't work for me. Here's what I did: Changed to my asterisk-1.4.23.1 directory Executed the wget / patch command from the link you provided make saw that res_features.so was recompiled Moved /usr/lib/asterisk/modules/res_features.so to res_features.so.old make install Confirmed that res_features.so was recreated in /usr/lib/asterisk/modules asterisk -r -- I never shut asterisk down module unload res_features.so module load res_features.so After this there was no change, it worked using the macro but using the Set(DYN... on the caller only. Thanks, All right. Let's continue this discussion on the bug report I opened. To start with, could you upload console output from an attempt at using the dynamic feature with my patch attached? For the console output, it would help if the verbose and debug levels were both set to at least 4. That way I can hopefully see what the problem is. Thanks. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/13/09 05:59:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)
It was with the patch applied, but after I restarted asterisk. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Friday, March 13, 2009 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4) David Ruggles wrote: I'm sorry, but it looks like it's working correctly now. I will update the bug if I am able to verify any problems. Thanks, Heh, no reason to be sorry for it working :) When you say it works now, was this with or without the patch applied? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/13/09 05:59:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trying to get sample applicationmap to work (*1.4)
I'm trying to actually use the example application map in features.conf: testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;tt-monkeys to the opposite channel I see the feature get registered at the CLI: == Registered Feature 'monkey' == Mapping Feature 'monkey' to app 'Playback(tt-monkeys)' with code '#9' But I'm unable to actually use it. This *doesn't* work: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) exten = 301,n,Dial(SIP/DavidR1) Anyone done this before and/or able to give me any suggestions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)
I don't really think that's a problem, because I'm able to use the other built in options: *1 to record; ## transfer (I changed this from a single pound) and there have been a couple times that I wouldn't hit them quickly enough. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Thursday, March 12, 2009 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4) David Ruggles wrote: I'm trying to actually use the example application map in features.conf: testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;tt-monkeys to the opposite channel I see the feature get registered at the CLI: == Registered Feature 'monkey' == Mapping Feature 'monkey' to app 'Playback(tt-monkeys)' with code '#9' But I'm unable to actually use it. This *doesn't* work: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) exten = 301,n,Dial(SIP/DavidR1) Anyone done this before and/or able to give me any suggestions? Thanks, I strongly suspect that you have fallen prey to the featuredigittimeout. Check your features.conf file for the featuredigittimeout option. By default, this is set to 500 ms. You probably want to increase this to something like 2000 ms. This option specifies the amount of time Asterisk should wait between DTMF presses when you are dialing a feature code. So in your case, I'm guessing that you pressed # but could not press 9 in time for Asterisk to recognize this input as part of the same feature. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/11/09 20:42:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)
Wow! Thanks! That's a very clear answer and completely understandable. Is this something I should open a bug report on? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Thursday, March 12, 2009 7:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4) David Ruggles wrote: I don't really think that's a problem, because I'm able to use the other built in options: *1 to record; ## transfer (I changed this from a single pound) and there have been a couple times that I wouldn't hit them quickly enough. Thanks, Ah, sorry about that. The featuredigittimeout burns so many people that it's pretty much a knee-jerk reaction on my part now to suggest that as a potential fix. To test out, I set up the same feature and gave it a try with a current subversion checkout of Asterisk 1.4. I placed a call from SIP/2001 to SIP/2000 and here's what I found. When SIP/2001 pressed #9, tt-monkeys played on SIP/2000's channel When SIP/2000 pressed #9, nothing happened. I tried modifying the features.conf line to have peer/callee instead of just peer and that caused neither side to successfully use the dynamic feature. There appears to be a bug which does not allow for the callee to use dynamic features. The problem appears to be that when DTMF is pressed, we try to interpret the presses to determine if there is a corresponding feature. The DYNAMIC_FEATURES variable has been set on the caller's channel, but has not been set on the callee's channel. As a result, we don't properly read the value of the DYNAMIC_FEATURES variable if the callee is the one to press DTMF. You can work around the bug, although it's not exactly optimal. What you can do is to modify your dialplan as follows: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) exten = 301,n,Dial(SIP/DavidR1,,M(dynamic_features)) [macro-dynamic_features] exten = s,1,Set(DYNAMIC_FEATURES=monkey) By doing this, the dynamic_features macro will be called on SIP/DavidR1 when he answers. This will allow for the DYNAMIC_FEATURES variable to be set on both channels so both sides can use the feature you have set. This is a bug, and so there needs to be action to fix it correctly. What I've suggested is just a workaround, but it should get you through your problem for now. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/12/09 10:38:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)
The patch doesn't work for me. Here's what I did: Changed to my asterisk-1.4.23.1 directory Executed the wget / patch command from the link you provided make saw that res_features.so was recompiled Moved /usr/lib/asterisk/modules/res_features.so to res_features.so.old make install Confirmed that res_features.so was recreated in /usr/lib/asterisk/modules asterisk -r -- I never shut asterisk down module unload res_features.so module load res_features.so After this there was no change, it worked using the macro but using the Set(DYN... on the caller only. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Thursday, March 12, 2009 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4) David Ruggles wrote: Wow! Thanks! That's a very clear answer and completely understandable. Is this something I should open a bug report on? Thanks, Nope, I've already got that taken care of. http://bugs.digium.com/view.php?id=14657 There's a patch there that I have tested and it works for me (TM). See if it works out for you, too. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/12/09 10:38:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID replacement
I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. Any suggestions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID replacement
Could you give me an example of how this would look in the dialplan? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Thursday, February 12, 2009 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID replacement On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID replacement
Some googling lead me to this: http://hans.fugal.net/blog/tag/astdb Which looks like it has an answer. Thanks all! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Thursday, February 12, 2009 12:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Caller ID replacement Could you give me an example of how this would look in the dialplan? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Nicholson Sent: Thursday, February 12, 2009 11:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID replacement On Thu, 2009-02-12 at 11:27 -0500, David Ruggles wrote: I'm working on building a pbx that will allow us to use our cellphones as extensions (to some extent) The dialout is working fine. What I would like to do is have an inbound cellphone call appear as if it were an extension. So right now if I call in from cell #9995551212 the caller id is 9995551212 but if I dial extension 30013 it will call cell #9995551212. I would like to change the caller id so 9995551212 is changed to 30013 on the inbound call. Doing one is simple enough, but I would like have an easy (more or less) way of setting up some global variables that link the cell phone #'s and extensions and have this done somewhat automagically. I would implement this using the a database (astdb or odbc) containing the mapping from cell number to extension. Then for each call that may need callerid modification, you can check the database for the proper mapping. With this method it is also easy to add new mappings. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1936 - Release Date: 02/05/09 11:34:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with 9133i config
'a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11' in 32000 ms (Method: REGISTER) --- SIP read from 192.168.0.11:5060 --- REGISTER sip:192.168.0.94 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869 Max-Forwards: 70 Content-Length: 0 To: myname sip:phone1@ From: myname sip:phone1@;tag=24b6354e352ab62 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER Contact: myname sip:pho...@192.168.0.11:5060;transport=udp Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.11 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@ Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:pho...@192.168.0.94 Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@;tag=as51ded290 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2f250e11 Content-Length: 0 Scheduling destruction of SIP dialog 'a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11' in 32000 ms (Method: REGISTER) --- SIP read from 192.168.0.11:5060 --- REGISTER sip:192.168.0.94 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869 Max-Forwards: 70 Content-Length: 0 To: myname sip:phone1@ From: myname sip:phone1@;tag=24b6354e352ab62 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER Contact: myname sip:pho...@192.168.0.11:5060;transport=udp Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Aastra 9133i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.11 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@ Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:pho...@192.168.0.94 Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.11:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.11:5060;branch=z9hG4bKcd3373869;received=192.168.0.11 From: myname sip:phone1@;tag=24b6354e352ab62 To: myname sip:phone1@;tag=as51ded290 Call-ID: a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11 CSeq: 1006065354 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2f250e11 Content-Length: 0 Scheduling destruction of SIP dialog 'a85fb0e71a3e7def4bd7d7f158933...@192.168.0.11' in 32000 ms (Method: REGISTER) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Troubleshoot in-bound DTMF over PRI
I have customer in the midwest that can't navigate my IVR Menus. I have two PRIs coming in to an Asterisk box. I enabled DTMF logging to the console and am able to see my own tones when I call in, however when this user calls in I see nothing. Is there a deeper level of debug I can do? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)
I'm trying to use the password entered with Authenticate to create dynamic meetme conferences with the following dial plan: exten = _XX18467,1,Authenticate(/etc/asterisk/meetme.pw|a) exten = _XX18467,n,MeetMe(CDR(accountcode)) ; 281-8467 However CDR(accountcode) is always being set to 1022 no matter what password is used. The passwords are stored in a file so they can easily be changed. Can anyone tell me what I'm doing wrong? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic meetme conference creation withAuthenticate (Asterisk 1.4.0)
I'm sorry for missing something so obvious! That was my problem.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Monday, March 24, 2008 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dynamic meetme conference creation
withAuthenticate (Asterisk 1.4.0)
On Monday 24 March 2008 10:56, David Ruggles wrote:
I'm trying to use the password entered with Authenticate to create dynamic
meetme conferences with the following dial plan:
exten = _XX18467,1,Authenticate(/etc/asterisk/meetme.pw|a)
exten = _XX18467,n,MeetMe(CDR(accountcode)) ; 281-8467
However CDR(accountcode) is always being set to 1022 no matter what
password is used. The passwords are stored in a file so they can easily be
changed.
Can anyone tell me what I'm doing wrong?
Well, the obvious part is that you're entering a conference NAMED
CDR(accountcode), not a conference with the value INSIDE CDR(accountcode).
Use MeetMe(${CDR(accountcode)}) to get that.
--
Tilghman
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Re: [asterisk-users] Dumb AGI question
Thank you sir! Am using verbose command now which works and I will take a look at the syslog. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Friday, November 16, 2007 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dumb AGI question On Fri, 16 Nov 2007, David Ruggles wrote: I did some simple AGI programming several months ago. I have a need to use one of those old programs and I'm having a stupid problem. I can't get output to display on the console. I'm sending it to stderr and I've got verbosity set to 10. I know I had it working before so I'm guessing I just forgot some piece of key information. Any suggestions? Jumping up on my syslog soapbox... Check out man syslog. If it fits your needs, logging to syslog is trivial. For example: syslog(LOG_ERR, The value of foo is %s, bar); If not, how about: ) Did you read all of the AGI environment from stdin? ) Why not use the agi command verbose? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-300 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dumb AGI question
I did some simple AGI programming several months ago. I have a need to use one of those old programs and I'm having a stupid problem. I can't get output to display on the console. I'm sending it to stderr and I've got verbosity set to 10. I know I had it working before so I'm guessing I just forgot some piece of key information. Any suggestions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Shared gsm files
Does anyone store gsm files on a shared server so multiple asterisk boxes can access the common gsm files? I want to do this so they can be updated easily, but wanted to make sure I wouldn't run in to any unforeseen problems. If anyone has done this could you tell me what you used and if you had any problems? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk IVR Performance
I have written a script that is executed using ExternalIVR(). I am running in to performance issues when I have four or more simultaneous calls running this script. I'm running on a P4 2.8 with 512M, all calls are GSM coming in over IAX from an asterisk box that acts as a switch and handles all PSTN interfaces. My question are these: Are there ways of optimizing ExternalIVRs? (maybe something like FastAGI) Right now I'm writing in a scripting language, would there be a performance gain from writing in a compiled language? I don't see any serious memory utilization and normally processor utilization is below 50% with spikes to 70% under load with four or five ExternalIVRs running. I will gladly provide any additional information that would aid in answering these questions. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what codecs for LAN
What is the best CODEC/Format combination to use? I've got several * boxes setup on a lan that are IVR servers. All the prompts are in GSM so I was using GSM thinking that it would prevent transcoding between the prompts and the voice channel. Is this an accurate assumption? Is there a better combination? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Wednesday, July 18, 2007 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] what codecs for LAN G711 is preferred if you wont face any bandwith limitation. That is why g729 is used on wan links. Voice quality should be better than g729 ans also less cpu load for asterisk. On 7/18/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have one more question about codec what codec i use for LAN setup G.729 or Alaw which is best for LAN setup caz some people told me G.729 is use for wan link not for lan caz it is cost effective so can anyone suggest me best codec for asterisk and SIP phone Rgds satish patel Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk mysql support
Issue: module load cdr_addon_mysql On the asterisk command line and post any error messages you receive Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Quintana Cruz Sent: Friday, June 01, 2007 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] asterisk mysql support Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. I'm using Xorcom's packages, cdr status shows: voip*CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend: cdr-custom it doesn't appear cdr_mysql. Any ideas? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] (no subject)
That made all the difference! Thanks again! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Wednesday, May 30, 2007 6:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] (no subject) Thanks; I have made the change and I will try it tomorrow! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cristian N. Bradiceanu Sent: Wednesday, May 30, 2007 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) Hi, Please take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g otchas iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up. in general section, add: iaxthreadcount = 200 in general section, add: iaxmaxthreadcount = 1000 Hope this helps. Regards, Cristian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
In both iax.conf files change [iax-trunk] to [tecinfo] the [name] in iax.conf is what is looked for when a connection is established and you're telling it to connect with tecinfo on the username= line HTH Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/ mailto:IAX2/192.168.253.20/[EMAIL PROTECTED]
tecinfo1/205)
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
username=tecinfo
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
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FW: [asterisk-users] Help with IAX
(missed one thing)
I have made some changed to your config:
extensions.conf from 192.168.253.21:
;
; Create an extension, 205, for trunk
;
exten = 205,1,Dial(IAX2/ mailto:IAX2/192.168.253.20/[EMAIL PROTECTED]
tecinfo1/205)
iax.conf from 192.168.253.21:
[tecinfo1]
type=peer
username=tecinfo2
secret=secret
host=192.168.253.20
---
extensions.conf from 192.168.253.20:
[iax-trunk]
exten = _205,1,Macro(voicemail,${E205})
iax.conf from 192.168.253.20:
[tecinfo2]
type=user
context=iax-trunk
secret=secret
host=192.168.253.21
Try this and respond with error messages (if any) from both systems
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
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[asterisk-users] (no subject)
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___|
| || | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk
boxes are using ztdummy for timing, they are all using IAX trunking.
My calls come in over Sip or Zap to asteriskm and are routed to one of the
asteriskN servers based on load. The routing is done by a small AGI script
that gets the current load from a monitoring machine and then changes the
priority. Dialplan snippet:
--- Snippet ---
exten = _X.,1,AGI(manager.agi)
exten = _X.,100,Dial(IAX2/asterisk1/${EXTEN})
exten = _X.,200,Dial(IAX2/asterisk2/${EXTEN})
exten = _X.,300,Dial(IAX2/asterisk3/${EXTEN})
exten = _X.,400,Dial(IAX2/asterisk4/${EXTEN})
exten = _X.,500,Dial(IAX2/asterisk5/${EXTEN})
--- Snippet ---
This works fine for a few calls. I'm using the SIPp package to generate a
10-25 simultaneous call load. Every once in a while I starting seeing loads
of error messages on AsteriskM's console:
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now UNREACHABLE! Time:
2
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
chan_iax2.c:909 __schedule_action: Out of idle IAX2 threads for scheduling!
chan_iax2.c:7054 socket_process: Peer 'asterisk4' is now REACHABLE! Time:
134
chan_iax2.c:6216 socket_read: Out of idle IAX2 threads for I/O, pausing!
That is just a small example, I may have 50-100 of these type of messages
scroll very quickly. If I give the system a minute everything goes back to
normal.
I would like some one who is very knowledgeable about IAX to assist me with
this problem. If someone knows a lot about IAX optimization and is willing
to work with me I would be willing to pay for their time.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
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RE: [asterisk-users] Help with IAX
Well this may not feel like progress, but it is. You no longer have an
authentication issue, you now have a routing issue. Could you attach a copy
of the extension.conf file on 192.168.253.21?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Malcom Kemp
Sent: Wednesday, May 30, 2007 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Help with IAX
From 192.168.253.20:
*CLI [May 30 14:39:03] NOTICE[6051]: chan_iax2.c:7146 socket_process:
Rejected connect attempt from 192.168.253.21, request '[EMAIL PROTECTED]' does
not exist
From 192.168.253.21:
[May 30 14:39:03] WARNING[28640]: chan_iax2.c:6959 socket_process: Call
rejected by 192.168.253.20: No such context/extension
I even changed the extension to take the pattern off:
exten = 205,1,Macro(voicemail,${E205})
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RE: [asterisk-users] (no subject)
Thanks; I have made the change and I will try it tomorrow! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cristian N. Bradiceanu Sent: Wednesday, May 30, 2007 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] (no subject) Hi, Please take a look at http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g otchas iax.conf The new threading model is great, but the default of 10 threads is way too low. Symptoms include total loss of audio until the channel is hung up. * in general section, add: iaxthreadcount = 200 * in general section, add: iaxmaxthreadcount = 1000 Hope this helps. Regards, Cristian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR Testing
Does anyone have any scripts or templates that could be used by an Asterisk box to call an IVR for stress testing? This task seems like it would be ideally suited to Asterisk and I wanted to see if anyone had already done this before I started trying to roll my own. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 threads
I've been running an asterisk box that is routing an average of 9 simultaneous calls between a PRI and five asterisk boxes via IAX. After having it run for about five hours I had a spate of these error messages: Out of idle IAX2 threads for I/O, pausing! And then they went away. The only references I can find on google for this error message are in the source. What does this mean and if it's a problem what should I be looking for to diagnose the cause? TIA Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax detection (Sangoma)
I have an FXS port and an FXO port on my Sangoma, can I detect inbound Fax calls on my FXO port and route them automatically to the FXS port (connected to a fax machine) while allowing normal voices to ring the main extension like normal? I looked through archive but didn't see this exact question addressed. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax detection (Sangoma)
Thanks for answering my question. I apologize for not looking harder, I'll look harder next time before asking the list. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ExternalIVR() Dialplan function and Festival
I ended up using text2wave to create a wav file and then added it to the prompt list and that worked. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Monday, March 19, 2007 5:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] ExternalIVR() Dialplan function and Festival Is there any way to use Festival from script called by the ExternalIVR() dialplan function? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExternalIVR() Dialplan function and Festival
Is there any way to use Festival from script called by the ExternalIVR() dialplan function? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Question (Asterisk 1.4 tarball)
I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the following message on asteriskm's cli: [Mar 13 11:34:33] NOTICE[6024]: chan_iax2.c:7840 __iax2_poke_noanswer: Peer 'asterisk1' is now UNREACHABLE! Time: 0 [Mar 13 11:36:13] WARNING[6029]: chan_iax2.c:3792 iax2_send: No private structure for packet? The warning repeats every 30 seconds, what am I doing wrong? Asteriskm config: **iax.conf** [general] bindaddr=192.168.0.160 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asterisk1] type=peer username=asteriskm auth=plaintext secret=asgard host=192.168.0.161 qualify=yes **extensions.conf** [general] [1ST-T1] exten = _X,1,AGI(rexx.agi) exten = 12345,1,Dial(IAX2/asterisk1/80483) exten = 12345,n,Hangup() Asterisk1 config: **iax.conf** [general] bindaddr=192.168.0.161 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asteriskm] type=user context=incoming-iax auth=plaintext secret=asgard host=192.168.0.160 qualify=yes trunk=yes **extensions.conf** [general] [incoming-iax] exten = _X,1,AGI(rexx.agi) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX2 Question (Asterisk 1.4 tarball)
The communication problem boiled down to iptables rules, but I'm still getting the No private structure for packet? error message. It doesn't seem to cause any problems and only occurs when an IAX2 peer has been unavailable for at least three minutes, but I would like to know why it happens if anyone knows. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Tuesday, March 13, 2007 11:38 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] IAX2 Question (Asterisk 1.4 tarball) I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the following message on asteriskm's cli: [Mar 13 11:34:33] NOTICE[6024]: chan_iax2.c:7840 __iax2_poke_noanswer: Peer 'asterisk1' is now UNREACHABLE! Time: 0 [Mar 13 11:36:13] WARNING[6029]: chan_iax2.c:3792 iax2_send: No private structure for packet? The warning repeats every 30 seconds, what am I doing wrong? Asteriskm config: **iax.conf** [general] bindaddr=192.168.0.160 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asterisk1] type=peer username=asteriskm auth=plaintext secret=asgard host=192.168.0.161 qualify=yes **extensions.conf** [general] [1ST-T1] exten = _X,1,AGI(rexx.agi) exten = 12345,1,Dial(IAX2/asterisk1/80483) exten = 12345,n,Hangup() Asterisk1 config: **iax.conf** [general] bindaddr=192.168.0.161 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asteriskm] type=user context=incoming-iax auth=plaintext secret=asgard host=192.168.0.160 qualify=yes trunk=yes **extensions.conf** [general] [incoming-iax] exten = _X,1,AGI(rexx.agi) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ACM question
I can telnet to the ACM on the local machine but I can't get to it from another machine I've been over the information about ACM at voip-info.org and haven't been able to figure out what I'm missing. I've included my manager.conf file and the error I'm getting from the other machine. Can anyone point out my problem? TIA Manager.conf: [general] displaysystemname = yes enabled = yes port = 5038 bindaddr = 0.0.0.0 [myuser] secret=mypass permit=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config Local System that works: [EMAIL PROTECTED] ~]# telnet 192.168.0.160 5038 Trying 192.168.0.160... Connected to 192-168-0-160.safedataisp.net (192.168.0.160). Escape character is '^]'. Asterisk Call Manager/1.0 ^] telnet quit Connection closed. [EMAIL PROTECTED] ~]# Remote system that doesn't work: [EMAIL PROTECTED] ~]# ping 192.168.0.160 PING 192.168.0.160 (192.168.0.160) 56(84) bytes of data. 64 bytes from 192.168.0.160: icmp_seq=1 ttl=64 time=0.298 ms 64 bytes from 192.168.0.160: icmp_seq=2 ttl=64 time=0.278 ms --- 192.168.0.160 ping statistics --- 2 packets transmitted, 2 received, 0% packet loss, time 999ms rtt min/avg/max/mdev = 0.278/0.288/0.298/0.010 ms [EMAIL PROTECTED] ~]# telnet 192.168.0.160 5038 Trying 192.168.0.160... telnet: connect to address 192.168.0.160: No route to host telnet: Unable to connect to remote host: No route to host [EMAIL PROTECTED] ~]# Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] ACM question
Thanks! That was the problem. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Monday, March 12, 2007 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ACM question On Mon, 12 Mar 2007, David Ruggles wrote: I can telnet to the ACM on the local machine but I can't get to it from another machine I've been over the information about ACM at voip-info.org and haven't been able to figure out what I'm missing. I've included my manager.conf file and the error I'm getting from the other machine. Can anyone point out my problem? Maybe iptables is getting in your way? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call load balancing
What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call load balancing
That's cool, but I doubt my systems could handle that same load ;) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Friday, March 09, 2007 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Call load balancing telco servers are Supermicro 2.8GHz P4, 1GB, Digium te410p with 2 PRI plugged in. application server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB, Digium te410p (timing only, all calls over IAX) database server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB No failures in over 2 years. On Fri, 9 Mar 2007, David Ruggles wrote: What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call load balancing
Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Can you give me a link to more information about how to use the management interface? I've been having a hard time trying to track down more advanced documentation and reference material. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call load balancing
Never mind I found it shortly after sending this :S Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, March 09, 2007 10:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Call load balancing Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Can you give me a link to more information about how to use the management interface? I've been having a hard time trying to track down more advanced documentation and reference material. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cdr_mysql compile question
I'm reading voip-info.org http://www.voip-info.org/wiki-Asterisk+cdr+mysql Sorry if this is a dumb question, but: It says I need mysql and mysql-devel to compile cdr_mysql, but I don't want mysql on my asterisk box I want to connect to a remote mysql server. Can I use mysqlclient and mysqlclient-devel? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cdr_mysql compile question
Nevermind, this was a dumb question :( Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, March 09, 2007 1:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Cdr_mysql compile question I'm reading voip-info.org http://www.voip-info.org/wiki-Asterisk+cdr+mysql Sorry if this is a dumb question, but: It says I need mysql and mysql-devel to compile cdr_mysql, but I don't want mysql on my asterisk box I want to connect to a remote mysql server. Can I use mysqlclient and mysqlclient-devel? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call load balancing
I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GSM cleanup (pops, clicks and static)
I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about 60 to 70 percent of the gsm files have some static or popping and clicking, on most of them it is in the silence at the end of the file. All that back story to ask this question: Are there any good utilities available for cleaning up gsm files? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Does Asterisk support DNIS?
Just to let everyone know. I restart my Asterisk box and ztcfg wouldn't run any more. I reran wancfg-zaptel and now everything is working correctly. It's picking up the DNIS digits without any problem. I still have to figure out ztcfg quits working every time I reboot though. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Does Asterisk support DNIS?
I played around the wink and rxwink settings. While increasing rxwink does delay the answer it still sees the DNIS digits individually. I changed the signalling to featd and now I get the following error: WARNING[27630]: chan_zap.c:5661 ss_thread: Got a non-Feature Group D input on channel 1. Assuming EM Wink instead Which I would expect, but the odd thing is that now it's seeing DNIS as a full extension. Going back to the em wink configuration: I found these settings for zapata.conf: prewink: Pre-wink time (default 50ms) preflash:Pre-flash time (default 50ms) wink:Wink time (default 150ms) flash: Flash time (default 750ms) start: Start time (default 1500ms) rxwink: Receiver wink time (default 300ms) rxflash: Receiver flashtime (default 1250ms) debounce:Debounce timing (default 600ms) Can anyone point me to some documentation that explains what these do for em_w signalling? Some of them seem obvious, but don't do what I would expect. With em wink the call answer should progress like this: Network goes off-hook PBX winks (goes off-hook) for 200ms Network sends DNIS as MF/DTMF tones inband PBX goes off-hook and answers. I would assume that wink means the same thing in zapata.conf, so I set it to 200. I also assumed that start meant answer, since there's no other option that seems to match, but changing it around didn't increase or decrease the amount of time it took to answer the call. As I said before changing rxwink did affect the amount of time, but I don't know what it's doing and it doesn't help Asterisk recognize the DNIS. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Does Asterisk support DNIS?
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference, I'm using a Sangoma A101 card. Asterisk sees each digit as a separate extension number so most of the dialplain suggestions offered so far won't work. I did try the Wait() function as was suggested. I tried it first in an s extension but this didn't work, it still gave the error: Unknown extension '1' in context '1st-T1' requested I then changed it to extension 1 and while it does seem to work (it doesn't try the other extensions) it seems like the DNIS is completely lost. As I said in my first post (although it may have been a little too abrasive) this configuration is very standard and so I find it hard to believe that Asterisk can't handle it. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Does Asterisk support DNIS?
Yuan (and Matt), Thanks for the reply, I'm sorry I kind of vented, I just got very frustrated with trying to configure Asterisk for what (in a proprietary PBX) is normally one of the easiest parts of configuration. With a wink start T1 the DNIS digits are transmitted in-band. The Network goes off hook, the PBX winks (goes off hook for 200ms) and then the network sends DNIS (and ANI if used) as DTMF tones, after the PBX gets the tones it answers the call (goes off hook). So you would tell the PBX to look for x number of digits and then after getting that number of digits it will answer the call. I have the Sangoma A101 configured for wink start, but I can't find anything that says how to specify the number DNIS digits to expect. If the PBX answers the call instead of just winking, the DTMF tones will be transmitted during the call which is what seems to be happening here. For more specific information a good overview of the wink start process can be found here: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080 1123bb.shtml#topic2a Can anyone tell me how to configure Asterisk to pickup the DNIS digits off a wink start T1? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 16, 2007 5:57 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Does Asterisk support DNIS? Matt already replied to your other posting of similar content. I'm also a bit confused. Do you mean you have observed that Asterisk is brought into the intended context, but start to react to digits in DNIS one after another? If so, can you estimate the interval Asterisk stays in each extension? If this is true, it seems to suggest that your provider is sending DNIS as a DTMF string after Asterisk has answered the call. Isn't this a bit weird? What does the card's manual say about DNIS (with wink start)? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A101 install problem
I just got a brand new A101 and am trying to install it in my test Asterisk box. The install went without a hitch. I followed the directions on the Sangoma Wiki: Wanpipe Asterisk Install http://sangoma.editme.com/wanpipe-linux-asterisk-install Wanpipe for Asterisk Configuration http://sangoma.editme.com/wanpipe-asterisk-configure It went perfect, no problems and Asterisk came up fine. I downed the box and moved it from my office to the computer room where I could hook it up to a test T1. I booted the box up and it looked like Zaptel wasn't installed. I went to the zaptel source directory and did a make clean, make install no errors, but now ztcfg -vvv shows this: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: E M (Default) (Slaves: 01) Channel 02: E M (Default) (Slaves: 02) Channel 03: E M (Default) (Slaves: 03) Channel 04: E M (Default) (Slaves: 04) Channel 05: E M (Default) (Slaves: 05) Channel 06: E M (Default) (Slaves: 06) Channel 07: E M (Default) (Slaves: 07) Channel 08: E M (Default) (Slaves: 08) Channel 09: E M (Default) (Slaves: 09) Channel 10: E M (Default) (Slaves: 10) Channel 11: E M (Default) (Slaves: 11) Channel 12: E M (Default) (Slaves: 12) Channel 13: E M (Default) (Slaves: 13) Channel 14: E M (Default) (Slaves: 14) Channel 15: E M (Default) (Slaves: 15) Channel 16: E M (Default) (Slaves: 16) Channel 17: E M (Default) (Slaves: 17) Channel 18: E M (Default) (Slaves: 18) Channel 19: E M (Default) (Slaves: 19) Channel 20: E M (Default) (Slaves: 20) Channel 21: E M (Default) (Slaves: 21) Channel 22: E M (Default) (Slaves: 22) Channel 23: E M (Default) (Slaves: 23) Channel 24: E M (Default) (Slaves: 24) Channel 25: FXS Kewlstart (Default) (Slaves: 25) 25 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) Any ideas what went wrong? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma A101 install problem
I reran the install and I had answered one question wrong. I think this fixed it. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 16, 2007 2:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sangoma A101 install problem I just got a brand new A101 and am trying to install it in my test Asterisk box. The install went without a hitch. I followed the directions on the Sangoma Wiki: Wanpipe Asterisk Install http://sangoma.editme.com/wanpipe-linux-asterisk-install Wanpipe for Asterisk Configuration http://sangoma.editme.com/wanpipe-asterisk-configure It went perfect, no problems and Asterisk came up fine. I downed the box and moved it from my office to the computer room where I could hook it up to a test T1. I booted the box up and it looked like Zaptel wasn't installed. I went to the zaptel source directory and did a make clean, make install no errors, but now ztcfg -vvv shows this: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: E M (Default) (Slaves: 01) Channel 02: E M (Default) (Slaves: 02) Channel 03: E M (Default) (Slaves: 03) Channel 04: E M (Default) (Slaves: 04) Channel 05: E M (Default) (Slaves: 05) Channel 06: E M (Default) (Slaves: 06) Channel 07: E M (Default) (Slaves: 07) Channel 08: E M (Default) (Slaves: 08) Channel 09: E M (Default) (Slaves: 09) Channel 10: E M (Default) (Slaves: 10) Channel 11: E M (Default) (Slaves: 11) Channel 12: E M (Default) (Slaves: 12) Channel 13: E M (Default) (Slaves: 13) Channel 14: E M (Default) (Slaves: 14) Channel 15: E M (Default) (Slaves: 15) Channel 16: E M (Default) (Slaves: 16) Channel 17: E M (Default) (Slaves: 17) Channel 18: E M (Default) (Slaves: 18) Channel 19: E M (Default) (Slaves: 19) Channel 20: E M (Default) (Slaves: 20) Channel 21: E M (Default) (Slaves: 21) Channel 22: E M (Default) (Slaves: 22) Channel 23: E M (Default) (Slaves: 23) Channel 24: E M (Default) (Slaves: 24) Channel 25: FXS Kewlstart (Default) (Slaves: 25) 25 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) Any ideas what went wrong? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DNIS on T1 channels
I installed a Sangoma card with the default install. I'm getting five digits of DNIS with each call. The T1 is setup ESF/B8ZS wink start. Each of the digits of the DNIS are being used for extensions in the context. I need a single extension that let me start an AGI script that can use the dnis. Can anyone point me in the right direction to do this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Asterisk support DNIS?
The subject pretty much says it all. Does Asterisk support DNIS, and if so, what kind of connection is required? (T1, PRI) I've got a wink start T1. I've read comments that say the DNIS will be seen as an extension, but I'm seeing each digit of the DNIS as a separate extension. So in my case I send DNIS of 12345, Asterisk will jump from extension 1 to extension 2 to extension 3 to extension 4 to extension 5. Only executing the first one or two lines in each. This is a PITA! And make absolutely no sense to me. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 card recommendation
I'm going to need to build a few Asterisk boxes that have dual and quad T1 interfaces. I knew Digium made T1 interface cards and on this list I heard about Sangoma so I did a quick search and found the hardware page at voip-info.org which lists several manufactures I didn't know about. All that leads to this question: I'll be using T1s in the USA. What experiences have you all had with different cards and what seems to work best? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI question
I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detect hang-up
I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: Friday, February 09, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Detect hang-up On Fri, 2007-02-09 at 15:31 -0500, David Ruggles wrote: I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! Maybe my zapata.conf can help you. I've one X100P working for almost 2 years :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
I've been doing some googling and I found references to using debug=1 with wctdm to see what's actually going on. It says this will be printed to the console. I'm running my * box headless in another room and sshing in to the box. I can't find where the debug out (if there is any) is going. Can any one point me in the right directions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 09, 2007 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Detect hang-up Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
By your post I can conclude that the console wctdm debugs to is the asterisk console. In that case I'm not getting anything from wctdm. I'm not using the safe_asterisk script I'm running asterisk -cvvv from the command line. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 09, 2007 5:45 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Detect hang-up From: David Ruggles [EMAIL PROTECTED] Date: Fri, 9 Feb 2007 16:43:41 -0500 I've been doing some googling and I found references to using debug=1 with wctdm to see what's actually going on. It says this will be printed to the console. I'm running my * box headless in another room and sshing in to the box. I can't find where the debug out (if there is any) is going. Can any one point me in the right directions? Two things to try. One is to simply start in console mode remotely (forget safe_asterisk). The other is to modify safe_asterisk script and disable console on ttyS9. Then when you start a remote console, STDERR will be there. Yuan Liu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 09, 2007 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Detect hang-up Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skutch AS-66 and an X100P
I finally got my X100P working and now I have a question. I have several Skutch phone line simulators. My X100P works as expected with both a POTS line and an analog PBX port, but when I use a phone line simulator it doesn't answer the line. The phone line simulator doesn't power the line until the phone set goes offhook. Asterisk shows the RED alarm and then the alarm clearing but never detects the ring. Any suggestions? TIA!!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New Issue
I'm still not seeing chan_zap in menu option three. I copied the source directories from /root/downloads/asterisk (where I had put them) to /usr/src/ and then did what you suggested below and I got the same result. I'm going to try make uninstalling all the packages deleted all source directories and starting over from the downloads. If you any other suggestions I'll do them. TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund Sent: Tuesday, February 06, 2007 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New Issue Try it like this: cd /usr/src/asterisk-1.4.0 make clean ./configure --with-zaptel=/usr/src/zaptel-1.4 make menuconfig make all make install David Ruggles wrote: Sorry about that I must have been in the wrong directory. I also have 1.4.0 and I tried it again and it worked. Chan_zap is not listed there, I'll start poking around and see if I stumble across anything. Do you know where the expected location is? I don't have a problem moving the source. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't get asterisk to compile chan_zap (was New Issue)
First, I didn't realize I hijacked another thread! Please accept my apologies. Now the problem: Asterisk isn't compiling chan_zap. chan_zap also doesn't appear in the list of channels when you make menuconfig I have read all the replies and specifically Cosmin's and Tzafrir's emails. zaptel.h is located in /usr/include/zaptel I also tried ./configure --with-zap=/usr/src/zaptel-1.4.0 which is the zaptel source and it didn't work either. I don't really know what else to try. I greatly appreciate any help TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Can't get asterisk to compile chan_zap (was NewIssue)
I had a typo in my last email. I meant --with-zaptel where I wrote --with-zap. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Can't get asterisk to compile chan_zap (was NewIssue)
I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't get asterisk to compile chan_zap (was NewIssue)
(I apologize if this is a dupe, but I never saw my first copy) I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Can't get asterisk to compile chan_zap (wasNewIssue)
I've been trying to snip message to keep them from getting too large, maybe I over did it. :) chan_zap.c is in /usr/src/asterisk-1.4.0/channels But doesn't show up in the list of channels in make menuconfig Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez Sent: Wednesday, February 07, 2007 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get asterisk to compile chan_zap (wasNewIssue) David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I didnt follow your original thread... chan_zap does not appear in menuselect? Does it exist in channels directory? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Can't get asterisk to compile chan_zap (wasNewIssue)
Menuselect-tree does have a member entry for chan_zap. I has two depend subnodes and one use subnode. The depends are: zaptel and tonezone The use is: pri (I've installed libpri also) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez Sent: Wednesday, February 07, 2007 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get asterisk to compile chan_zap (wasNewIssue) David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users check too that menuselect-tree has an entry for chan_zap (it's in source root) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Can't get asterisk to compile chan_zap(wasNewIssue)
that I have! :) Have a single X100P in the system and ztcfg configures the board no problem. zttool confirms the board is there and shows RED when the phone line is removed and OK when the phone line is plugged in. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Wednesday, February 07, 2007 2:02 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Can't get asterisk to compile chan_zap(wasNewIssue) From: David Ruggles [EMAIL PROTECTED] Date: Wed, 7 Feb 2007 12:15:37 -0500 I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? At least one version of Asterisk (1.4?) requires correct kernel driver configuration before compilation. Have you done ztconfig and stuff? Yuan Liu Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New user question (X100P)
Thanks for the help! ZT_CHANCONFIG failed on channel 1: No such device or address (6) This error message is specifically from running ztcfg. What do you have in /proc/zaptel/* ? What in /etc/zaptel.conf ? The difference between the two is the immediate cause for the error. In /proc/zaptel I have a single file named 1 In /etc/zaptel.conf I have the following lines: fxsks=1-2 loadzone=us defaultzone=us What do you see in the kernel logs e.g: dmesg| tail ) after loading the module wcfxo ? I did the following: # rmmod wcfxo # modprobe wcfxo ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo # dmesg | tail Failed to initailize DAA, giving up... Wcfxo: probe of :01:07.0 failed with error -5 Failed to initailize DAA, giving up... Wcfxo: probe of :01:08.0 failed with error -5 Using lspci I verified that 01:07.0 and 01:08.0 are the X100P What linux distribution do you use, anyway? Fedora Core 6 Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New user question (X100P) (ADDENDUM)
I read all the other replies and want to thank everyone!! I think most of what other people mentioned is answered below. I'm only doing this to test Asterisk. I will be using T1 cards when I start putting Asterisk in production. I've got several Ad-tran TSU 600s and 120s that I can use for analog access when I get to that point. All my prior work was using custom software on NMS AG8s, AGT1s and AG4000s which aren't supported by Asterisk so I wanted to use the X100Ps for proof-of-concept and to get familiar with Asterisk before buying the digital interface cards. Thanks for the help! ZT_CHANCONFIG failed on channel 1: No such device or address (6) This error message is specifically from running ztcfg. What do you have in /proc/zaptel/* ? What in /etc/zaptel.conf ? The difference between the two is the immediate cause for the error. In /proc/zaptel I have a single file named 1 In /etc/zaptel.conf I have the following lines: fxsks=1-2 loadzone=us defaultzone=us What do you see in the kernel logs e.g: dmesg| tail ) after loading the module wcfxo ? I did the following: # rmmod wcfxo # modprobe wcfxo ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo # dmesg | tail Failed to initailize DAA, giving up... Wcfxo: probe of :01:07.0 failed with error -5 Failed to initailize DAA, giving up... Wcfxo: probe of :01:08.0 failed with error -5 Using lspci I verified that 01:07.0 and 01:08.0 are the X100P What linux distribution do you use, anyway? Fedora Core 6 Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New user question (X100P) (ADDENDUM)
One other note, the output of lspci for the two X100Ps is: 01:07.0 Communication controller: Motorola Wildcard X100P 01:08.0 Communication controller: Motorola Wildcard X100P Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New user question (X100P)
What is the output of cat /proc/zaptel/1 ? (My guess: ztdummy) Correct Do you have two cards? Hmm... you do seem to have two of those... Either a defective hardware or misunderstandings with the PCI bus. For instance, on one system I know, I had to pass the kernel parameter 'pci=noacpi' at boot in order for it to identify those cards (as well as TDM400P cards). Tried this and it didn't make any difference. What kernmel version? 2.6.19-1.2895 Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New user question (X100P) SOLVED!!!
Thanks everyone! I removed the extra X100P and tried the remaining X100P in both PCI slots and it works in one and doesn't work in the other. I really only need one for early testing so this is good. Thanks again! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Issue
Now that ztcfg is working correctly I can't seem to get asterisk to answer a call. I did the make install and make samples so I have a lot of configuration files that I know nothing about. Here is contents of zapata.conf [trunkgroups] [channels] context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes channel = 1 And the contents of extensions.conf [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [incoming] exten = s,1,Answer() exten = s,2,Echo() This from TFOT, the general and globals sections of extensions came from the sample. I started Asterisk using asterisk -cvvv and it doesn't seem to have any errors, but I can't find where it parses zapata.conf. I do see it parsing extensions.conf What should I do? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New Issue
I'm missing chan_zap.so, I'm going to make and make install again as per: http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Tuesday, February 06, 2007 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] New Issue Now that ztcfg is working correctly I can't seem to get asterisk to answer a call. I did the make install and make samples so I have a lot of configuration files that I know nothing about. Here is contents of zapata.conf [trunkgroups] [channels] context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes channel = 1 And the contents of extensions.conf [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [incoming] exten = s,1,Answer() exten = s,2,Echo() This from TFOT, the general and globals sections of extensions came from the sample. I started Asterisk using asterisk -cvvv and it doesn't seem to have any errors, but I can't find where it parses zapata.conf. I do see it parsing extensions.conf What should I do? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New Issue
Well that didn't work. I still don't have a zap channel driver. What else can I try? TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Tuesday, February 06, 2007 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] New Issue I'm missing chan_zap.so, I'm going to make and make install again as per: http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Tuesday, February 06, 2007 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] New Issue Now that ztcfg is working correctly I can't seem to get asterisk to answer a call. I did the make install and make samples so I have a lot of configuration files that I know nothing about. Here is contents of zapata.conf [trunkgroups] [channels] context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes channel = 1 And the contents of extensions.conf [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [incoming] exten = s,1,Answer() exten = s,2,Echo() This from TFOT, the general and globals sections of extensions came from the sample. I started Asterisk using asterisk -cvvv and it doesn't seem to have any errors, but I can't find where it parses zapata.conf. I do see it parsing extensions.conf What should I do? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New Issue
Thanks for the reply, but when I go to the asterisk source directory and issue make menuconfig I get: make: *** No rule to make target `menuconfig'. Stop. The source I have is the latest tar file from the astrisk site. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund Sent: Tuesday, February 06, 2007 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New Issue Let's see if I remember this, it gave me a bit of trouble as well. *after* you made sure you've got the zaptel driver in order, go to the src folder for asterysk and issue make menuconfig. Go to 3 and see if you have the chan_zap listed there and with [*] prefix. If it's not listed it's because you've got the zaptel driver sources in an unexpected location. You'll need to manually specify the location of you zaptel driver (don't remember how) and then re-issue the make menuconfig. This time you'll see the zap chan driver available as an option at 3. Now exist the make menuconfig SAVING your changes. I didn't save (since I didn't make any changes) and my channel driver didn't build. I tried it again, saved the config and it worked. Hope someone can fell the missing bit, the way to tell make where to find the zaptel source files. David Ruggles wrote: Well that didn't work. I still don't have a zap channel driver. What else can I try? TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Tuesday, February 06, 2007 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] New Issue I'm missing chan_zap.so, I'm going to make and make install again as per: http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Tuesday, February 06, 2007 3:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] New Issue Now that ztcfg is working correctly I can't seem to get asterisk to answer a call. I did the make install and make samples so I have a lot of configuration files that I know nothing about. Here is contents of zapata.conf [trunkgroups] [channels] context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes channel = 1 And the contents of extensions.conf [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1; MSD digits to strip (usually 1 or 0) [incoming] exten = s,1,Answer() exten = s,2,Echo() This from TFOT, the general and globals sections of extensions came from the sample. I started Asterisk using asterisk -cvvv and it doesn't seem to have any errors, but I can't find where it parses zapata.conf. I do see it parsing extensions.conf What should I do? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] New Issue
Sorry about that I must have been in the wrong directory. I also have 1.4.0 and I tried it again and it worked. Chan_zap is not listed there, I'll start poking around and see if I stumble across anything. Do you know where the expected location is? I don't have a problem moving the source. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund Sent: Tuesday, February 06, 2007 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New Issue I'v got Asterisk 1.4.0 and it understands make menuconfig. Is your version older or newer? If it's older, maybe you can try the newer one. If it's newer - I'm out of ideas. David Ruggles wrote: Thanks for the reply, but when I go to the asterisk source directory and issue make menuconfig I get: make: *** No rule to make target `menuconfig'. Stop. The source I have is the latest tar file from the astrisk site. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New user question (X100P)
I'm trying to set up a simple test box to start developing with Asterisk. I've got a Dell GX150 with two X100P cards. I've downloaded, printed out and read through most of TFOT. I've also done a lot of Internet searching. I'm getting this error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) Based on the searching I've done, it seems like the problem must be shared IRQ issues. I've gone in the BIOS and disabled everything I can but I can't stop the sharing completely. One X100P is shared with the built-in video and the other X100P is shared with a serial controller. (I disabled both serial ports) My question is this: (in three parts) 1) Are my research and assumptions accurate? Does this seem to be an IRQ issue? 2) If I have to build another system to prevent the IRQ problem, can anyone recommend hardware (just for a simple test box right now) 3) Is it worth keeping the X100Ps or should I get a TDM400P like used in TFOT, or something else? (again just for testing) If there are other things I should check first please let me know!! TIA! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
