Re: [asterisk-users] Libtonezone
You could read the source code, but based on it's name I would say it is a library responsible for zone specific tone generation. Many parts of the world have different tone patterns than the U.S. and Asterisk is used worldwide. A better question is, why are you concerned by it? From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Joseph L. Casale [jcas...@activenetwerx.com] Sent: Sunday, March 28, 2010 9:13 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Libtonezone Trying to find out what the libtonezone shared object built with dahdi-tools is for, the default dahdi package installation from the Digium repo's pull it in, so when is it needed? Thanks, jlc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pri CLI command not available
This is often caused by the dahdi module not loading, check /var/log/asterisk/messages for the reason, or better yet, from the cli load the module manually and see the error in real time. If I had to guess I would say it is a configuration error. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Merkel (Mail Lists) Sent: Thursday, January 21, 2010 1:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pri CLI command not available I am in the process of trying to terminate a PRI into a new * server. The server has an old T100P T1/PRI card in it. I have compiled the following on Centos 5.4. dahdi-linux-complete-2.2.1+2.2.1 libpri-1.4.10.2 asterisk-1.4.29 Everything seems to have compiled fine. DAHDI reports Found a Wildcard: Digium Wildcard T100P T1/PRI on bootup. Dahdi_tool shows that the span is up and active with no alarms however the phone company is not seeing the trunkgroup going into service. I was wanting to take a look at the PRI debugs but for some reason the CLI pri option is not available. I libpri compiled without any issues prior to compiling asterisk. What would cause the pri debug commands to not be available in the CLI? = Eric Merkel ejmerkel.li...@gmail.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] CID not working.
You need to wait at least 1 second on an incoming POTS line for CID info, add a wait(1) as the first step on incoming connections. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arun Sasidhar Sent: Wednesday, December 30, 2009 7:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CID not working. Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing Unknown when there is an incoming call. My log file showing this while an incoming call on PSTN line: tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0 [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Starting simple switch on 'DAHDI/1-1' [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:1] Set(DAHDI/1-1, __FROM_DID=s) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:2] Gosub(DAHDI/1-1, app-blacklist-check|s|1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:1] LookupBlacklist(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:2] GotoIf(DAHDI/1-1, 0?blacklisted) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:3] Set(DAHDI/1-1, CALLED_BLACKLIST=1) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@app-blacklist-check:4] Return(DAHDI/1-1, ) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:3] ExecIf(DAHDI/1-1, 1 |Set|CALLERID(name)=) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:4] Set(DAHDI/1-1, FAX_RX=disabled) in new stack [Dec 30 06:36:16] VERBOSE[2715] logger.c: -- Executing [...@from-pstn:5] Set(DAHDI/1-1, __CALLINGPRES_SV=allowed_not_screened) in new stack My chan_dahdi.conf file is as like this. vim /etc/asterisk/chan_dahdi.conf [channels] language=en hanguponpolarityswitch=yes answeronpolarityswitch=yes busydetect=yes busycount=3 callprogress=yes callerid=asreceived immediate=yes cidsignalling=dtmf cidstart=polarity ;cidstart=ring useincomingcalleridonzaptransfer=yes ;cidsignalling=bell ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf ; XTDM20B Port #1,2 plugged into PSTN ;AMPLABEL:Channel %c - Button %n Please help me for fixing this issue. I am from India. Regards, Aruns ___ -- 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] CDR
If asterisk enters the answered state at any point in the call, then the call disposition becomes answered. Thank you and have a nice day, Anthony Francis From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Szasz Szabolcs Sent: Tuesday, December 29, 2009 12:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] CDR Hi, How does Asterisk CDR work? How can I have in CDR records calls without BYE message? I checked my wireshark traces and some calls has no BYE messages, but they appears in CDR as answered call. Thanks Szabolcs Szasz ___ -- 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] FAX for Asterisk
Where do you get FFA? I have not seen this, what is the minimum version of Asterisk that you need? Sorry about the questions. Thank you and have a nice day, Anthony Francis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: Thursday, December 17, 2009 8:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FAX for Asterisk Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI module load res_f res_fax.so res_features.so res_fax_digium.so astbh00*CLI module load res_f Help! :) ___ -- 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] What version of libpri and zaptel work best with 1.4.24
Hello all, I am trying to use asterisk 1.4.24 so that I can get app_rxfax working, I installed it, along with the versions of libpri and zaptel that had release dates closest to the release date of 1.4.24, however, I now have a problem where outbound dialing now fails, cause 99 on the PRI. Does anyone know which version of libpri and zaptel I should be using? I cannot find a good reference to this. Thank you and have a nice day, Anthony Francis ___ -- 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] DTMF
Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: I went ahead and switched to SIP just for grins, and made sure dtmfmode=rfc2833 is in the peer config on both sides and in the entry for the phone. So now it is: polycom501---SIP/ulaw---ast1---SIP/g729---ast2---IAX/ulaw---ITSP A bit more information. ast1 is running 1.4.23.1 and I noticed a debug line in rtp.c: if (rtpdebug || option_debug 2) ast_log(LOG_DEBUG, - RTP 2833 Event: %08x (len = %d)\n, event, len); So I set debug to 10 and caught this line: [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 0002 (len = 4) So I guess that proves that from the phone to ast1 RFC2833 is in effect (I did actually press the digit '2', which I assume is the event code above?). I tried to do the same on ast2, which is running 1.4.22.1, and with debug set to 10 I did *not* get this message, which makes me think that RCF2833 is NOT in effect for the trunk between ast1 and ast2. Is that reasonable? The main problem turned out to be at my ITSP, and is now resolved. The question remains for me, though, how to interpret the debug lines I was able to catch (or not) above. How do you really know if RFC2833 signalling is being received? I caught the debug message on ast1 but not on ast2. I am using ulaw between ast2 and the ITSP, and I am now wondering if the DTMF is being sent inband on that last leg since I could not catch the debug messages on ast2. Perhaps what they did to fix on their end is simply remove compression between themselves and the PSTN. I would really like a concrete method of verifying that DTMF signalling is being sent out of band on my outbound IAX link. Any ideas? Thanks, j You are correct, not seeing that means that the signaling was either in the audio stream (which doesn't survive compression) or it was sent in the sip signaling. However one must also note that your ITSP's gateway may have been having problems with their DTMF detection on their PRI's. Anthony ___ -- 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] DTMF
Anthony Francis wrote: Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: I went ahead and switched to SIP just for grins, and made sure dtmfmode=rfc2833 is in the peer config on both sides and in the entry for the phone. So now it is: polycom501---SIP/ulaw---ast1---SIP/g729---ast2---IAX/ulaw---ITSP A bit more information. ast1 is running 1.4.23.1 and I noticed a debug line in rtp.c: if (rtpdebug || option_debug 2) ast_log(LOG_DEBUG, - RTP 2833 Event: %08x (len = %d)\n, event, len); So I set debug to 10 and caught this line: [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 0002 (len = 4) So I guess that proves that from the phone to ast1 RFC2833 is in effect (I did actually press the digit '2', which I assume is the event code above?). I tried to do the same on ast2, which is running 1.4.22.1, and with debug set to 10 I did *not* get this message, which makes me think that RCF2833 is NOT in effect for the trunk between ast1 and ast2. Is that reasonable? The main problem turned out to be at my ITSP, and is now resolved. The question remains for me, though, how to interpret the debug lines I was able to catch (or not) above. How do you really know if RFC2833 signalling is being received? I caught the debug message on ast1 but not on ast2. I am using ulaw between ast2 and the ITSP, and I am now wondering if the DTMF is being sent inband on that last leg since I could not catch the debug messages on ast2. Perhaps what they did to fix on their end is simply remove compression between themselves and the PSTN. I would really like a concrete method of verifying that DTMF signalling is being sent out of band on my outbound IAX link. Any ideas? Thanks, j You are correct, not seeing that means that the signaling was either in the audio stream (which doesn't survive compression) or it was sent in the sip signaling. However one must also note that your ITSP's gateway may have been having problems with their DTMF detection on their PRI's. Anthony Also, to determine if you are sending DTMF out of band (as part of IAX signalling) do iax2 debug peer connection name in the CLI. You will see when it creates DTMF events. ___ -- 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] inbound filed
Bayardo Sanchez wrote: tollfree calls was working fine but stopped working without any reason Oh, there's a reason. ___ -- 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] IPkall
SIP wrote: IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. N. Dean Collins wrote: Does IPKALL still exist? I am after a free SIP trunk – who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). The sign up link doesn't work. ___ -- 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] Cdr problem
Tilghman Lesher wrote: On Monday 09 March 2009 01:28:49 pm Anthony Francis wrote: Tilghman Lesher wrote: On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote: hi, I'm working with asterisk on a project and I found a problem with cdr_odbc. As we know, after answering each call a cdr event is raised which is saved in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some information, including start_time and end_time is given by cdr event but the problem is that these two information(start_time and end_time) is not getting save in cdr_odbc. I checked the source code and I found that by default it's not doing so. I need to query these two information, start time and end time, from cdr_odbc and I need your help. thanks You are partially incorrect. The start time is indeed stored in the CDR, although the column name is 'calldate'. As for the end time, it can be derived by adding 'duration' (which is in whole seconds) to the 'calldate' column. Another solution that allows for retrieving both columns with their native names (or completely different names, whatever you map it to) is to use cdr_adaptive_odbc in 1.6.0 and higher. I have often thought, wouldn't it be better if the cdr config files allowed you to specify column names i.e. calldate = callstart_datetime Or whatever, the basic format being asteriskfieldname = db columnname. Just an idea.. Which is how cdr_adaptive_odbc already works. ;-) Yeah I haven't moved to 1.6 yet :(. ___ -- 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] Cdr problem
Tilghman Lesher wrote: On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote: hi, I'm working with asterisk on a project and I found a problem with cdr_odbc. As we know, after answering each call a cdr event is raised which is saved in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some information, including start_time and end_time is given by cdr event but the problem is that these two information(start_time and end_time) is not getting save in cdr_odbc. I checked the source code and I found that by default it's not doing so. I need to query these two information, start time and end time, from cdr_odbc and I need your help. thanks You are partially incorrect. The start time is indeed stored in the CDR, although the column name is 'calldate'. As for the end time, it can be derived by adding 'duration' (which is in whole seconds) to the 'calldate' column. Another solution that allows for retrieving both columns with their native names (or completely different names, whatever you map it to) is to use cdr_adaptive_odbc in 1.6.0 and higher. I have often thought, wouldn't it be better if the cdr config files allowed you to specify column names i.e. calldate = callstart_datetime Or whatever, the basic format being asteriskfieldname = db columnname. Just an idea.. Anthony Francis ___ -- 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] Bounty- CDR Bug Fix
Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a patch way back for 1.2 that allowed all queue log events to sh,ow up in the AMI, just haven't had time to make a new version for 1.6. Maybe this time I can get the patch in trunk and it will always be there. Robert Broyles wrote: Problem is, without going to 1.6, I can't get the queue log or events posted to MySQL in realtime. There used to be a patch out there for queue_log, but it doesn't work with versions 1.4.21 or higher. -- Regards, Robert Broyles Anthony Francis wrote: Robert Broyles wrote: I saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a bounty out on it. http://bugs.digium.com/view.php?id=13691 I would not recommend using CDR's for queue data, instead I use the queue events, or at a minimum the queue log. ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.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] Bounty- CDR Bug Fix
Hmm, Yeah something that writes direct to MySQL would not have made it into trunk, I took the route of using the queue event flag to also turn on and off sending of all queue events normally written to the log to also go to the AMI. I also have a perl script that listens to the AMI for events and puts them in a db. Robert Broyles wrote: The patch I was referring to is: http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging It doesn't work for the current SVN 1.4 -- Regards, Robert Broyles Anthony Francis wrote: Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a patch way back for 1.2 that allowed all queue log events to sh,ow up in the AMI, just haven't had time to make a new version for 1.6. Maybe this time I can get the patch in trunk and it will always be there. Robert Broyles wrote: Problem is, without going to 1.6, I can't get the queue log or events posted to MySQL in realtime. There used to be a patch out there for queue_log, but it doesn't work with versions 1.4.21 or higher. -- Regards, Robert Broyles Anthony Francis wrote: Robert Broyles wrote: I saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a bounty out on it. http://bugs.digium.com/view.php?id=13691 I would not recommend using CDR's for queue data, instead I use the queue events, or at a minimum the queue log. ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.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] AEL2: If-then-else not permitted in Switch-Case
Philipp Kempgen wrote: Courier mail server at exa.billmerriam.com schrieb: This is a delivery status notification from exa.billmerriam.com, running the Courier mail server, version 0.54.1. The original message was received on Wed, 04 Mar 2009 09:10:55 -0500 from localhost (localhost [127.0.0.1]) --- UNDELIVERABLE MAIL Your message to the following recipients cannot be delivered: li...@billmerriam.com: yocto.billmerriam.com [68.209.186.200]: STARTTLS 500 couriertls: connect: Connection reset by peer li...@billmerriam.com, please fix your mail server. I sent the message to the asterisk-users mailing list and - sorry to say - I don't care if it was delivered to you or not. Thanks, Philipp Kempgen ...and so you replied to it? I mean if he didn't get the original copy, he sure isn't going to get your terse reply. The rest of us however -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] Bounty- CDR Bug Fix
Robert Broyles wrote: I saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a bounty out on it. http://bugs.digium.com/view.php?id=13691 I would not recommend using CDR's for queue data, instead I use the queue events, or at a minimum the queue log. -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] Cisco IP Phone 7940G.
David Gibbons wrote: I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment. I'll provide SEPMAC.cnf.xml's if requested off-list. --Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, February 13, 2009 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phone 7940G. Catalin S. wrote: hey finally i did it. I upgraded the firmware to the latest sip firmware and now i have the another problem. The requested files are the following: ---///--- Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SEP00141CAA4B4C.cnf.xml to 192.168.1.3:51252 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIP00141CAA4B4C.cnf to 192.168.1.3:51253 Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving SIPDefault.cnf to 192.168.1.3:51254 ---///--- I made my own sip configuration in SIP00141CAA4B4C.cnf where 00141CAA4B4C is the mac address of phone, but i don't know what to write in CTLSEP00141CAA4B4C.tlv, Create an empty file and it will be happy. At least that has been my experience with my 7960. Others can probably provide a sample of the remaining files. So far I have been unable to go beyond version 7 firmware, as it is unhappy with the XML file when trying to move to version 8. John Novack -- Dog is my co-pilot On a similar subject, I have been able to get a 7961 to switch to a SIP firmware, has anyone had any luck with this? -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] AgentCallBackLogin no longer works after installing asterisk 1.6
oumar ndiaye wrote:
Thanks all for your responses.
I am not sure I know every thing AgentCallBackLogin is capable. I
don't know either if I have to have all the functions offered by
AgentCallBackLogin. All I need is a way to allow call takers to login
and before they can take calls. How is this done today in 1.6.
Thanks.
On Fri, Feb 6, 2009 at 7:40 PM, Philipp Kempgen
philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote:
Rob Hillis schrieb:
...except that Macros are now deprecated and will most likely be
removed
in 1.8.
Robert Broyles wrote:
Looks like using a Macro and the 'M' Dial() option is the way
to go for
now if you need the answer confirmation.
Use U() and Gosubs then!
Philipp Kempgen
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Tel: +1-919-291-8742
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So here is what I have come up with to solve the problem and be light on
resources,
This takes any file that already exists and symbolically links it to
what ever file you specify, this way the caller doesn't ever have to
listen to a prompt to record their name, then when the agent answers,
their only real option is to hit 1 to answer the call. I tested this and
it seems to work.
exten = _X.,1,System(ln -sf /var/lib/asterisk/sounds/vm-Work.gsm
/var/lib/asterisk/sounds/priv-callerintros/${IF($[ ${CALLERID(num)} !=
]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.gsm)
exten = _X.,n,Set(AGENT_LOC=${DB(rockynet/agent/${EXTEN})})
exten = _X.,n,Dial(Local/${agent_l...@rockynet-support,20,trp)
--
Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
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Re: [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6
Anthony Francis wrote:
oumar ndiaye wrote:
Thanks all for your responses.
I am not sure I know every thing AgentCallBackLogin is capable. I
don't know either if I have to have all the functions offered by
AgentCallBackLogin. All I need is a way to allow call takers to login
and before they can take calls. How is this done today in 1.6.
Thanks.
On Fri, Feb 6, 2009 at 7:40 PM, Philipp Kempgen
philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote:
Rob Hillis schrieb:
...except that Macros are now deprecated and will most likely be
removed
in 1.8.
Robert Broyles wrote:
Looks like using a Macro and the 'M' Dial() option is the way
to go for
now if you need the answer confirmation.
Use U() and Gosubs then!
Philipp Kempgen
So here is what I have come up with to solve the problem and be light on
resources,
This takes any file that already exists and symbolically links it to
what ever file you specify, this way the caller doesn't ever have to
listen to a prompt to record their name, then when the agent answers,
their only real option is to hit 1 to answer the call. I tested this and
it seems to work.
exten = _X.,1,System(ln -sf /var/lib/asterisk/sounds/vm-Work.gsm
/var/lib/asterisk/sounds/priv-callerintros/${IF($[ ${CALLERID(num)} !=
]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.gsm)
exten = _X.,n,Set(AGENT_LOC=${DB(rockynet/agent/${EXTEN})})
exten = _X.,n,Dial(Local/${agent_l...@rockynet-support,20,trp)
Oh and here is the login / out toggle exten I came up with:
exten = *77,1,VMAuthenticate(@rockynet|)
exten =
*77,n,AddQueueMember(rockynet-service|local/${auth_mailb...@rockynet-agents)
exten = *77,n,Read(AGENT_LOC|agent-newlocation)
exten = *77,n,Set(DB(rockynet-1000/${AUTH_MAILBOX})=${AGENT_LOC})
exten = *77,n,Goto(*77-${AQMSTATUS}|1)
exten = *77-ADDED,1,Background(agent-loginok)
exten = *77-ADDED,n,Hangup()
exten =
*77-MEMBERALREADY,1,RemoveQueueMember(rockynet-service|local/${auth_mailb...@rockynet-agents)
exten =
*77-MEMBERALREADY,n,Set(oldvar=${DB_DELETE(rockynet-/agent/${AUTH_MAILBOX})})
exten = *77-MEMBERALREADY,n,Background(agent-loggedoff)
exten = *77-MEMBERALREADY,n,Hangup()
This is based on that blog post but uses less extensions, I personally
prefer toggle style queue controls.
--
Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
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Re: [asterisk-users] AgentCallBackLogin no longer works after installing asterisk 1.6
Anthony Francis wrote:
Anthony Francis wrote:
oumar ndiaye wrote:
Thanks all for your responses.
I am not sure I know every thing AgentCallBackLogin is capable. I
don't know either if I have to have all the functions offered by
AgentCallBackLogin. All I need is a way to allow call takers to login
and before they can take calls. How is this done today in 1.6.
Thanks.
On Fri, Feb 6, 2009 at 7:40 PM, Philipp Kempgen
philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote:
Rob Hillis schrieb:
...except that Macros are now deprecated and will most likely be
removed
in 1.8.
Robert Broyles wrote:
Looks like using a Macro and the 'M' Dial() option is the way
to go for
now if you need the answer confirmation.
Use U() and Gosubs then!
Philipp Kempgen
So here is what I have come up with to solve the problem and be light on
resources,
This takes any file that already exists and symbolically links it to
what ever file you specify, this way the caller doesn't ever have to
listen to a prompt to record their name, then when the agent answers,
their only real option is to hit 1 to answer the call. I tested this and
it seems to work.
exten = _X.,1,System(ln -sf /var/lib/asterisk/sounds/vm-Work.gsm
/var/lib/asterisk/sounds/priv-callerintros/${IF($[ ${CALLERID(num)} !=
]?${CALLERID(num)}:NOCALLERID_${EXTEN}${CUT(CHANNEL,/,1)}=${CUT(CHANNEL,/,2)})}.gsm)
exten = _X.,n,Set(AGENT_LOC=${DB(rockynet/agent/${EXTEN})})
exten = _X.,n,Dial(Local/${agent_l...@rockynet-support,20,trp)
Oh and here is the login / out toggle exten I came up with:
exten = *77,1,VMAuthenticate(@rockynet|)
exten =
*77,n,AddQueueMember(rockynet-service|local/${auth_mailb...@rockynet-agents)
exten = *77,n,Read(AGENT_LOC|agent-newlocation)
exten = *77,n,Set(DB(rockynet-1000/${AUTH_MAILBOX})=${AGENT_LOC})
exten = *77,n,Goto(*77-${AQMSTATUS}|1)
exten = *77-ADDED,1,Background(agent-loginok)
exten = *77-ADDED,n,Hangup()
exten =
*77-MEMBERALREADY,1,RemoveQueueMember(rockynet-service|local/${auth_mailb...@rockynet-agents)
exten =
*77-MEMBERALREADY,n,Set(oldvar=${DB_DELETE(rockynet-/agent/${AUTH_MAILBOX})})
exten = *77-MEMBERALREADY,n,Background(agent-loggedoff)
exten = *77-MEMBERALREADY,n,Hangup()
This is based on that blog post but uses less extensions, I personally
prefer toggle style queue controls.
While this looked like a solution at first, it appears it is not as the
called party picking up the line (them or their vm) does return an
answered state to queue. So the question then is, does using the U
option in 1.6 have the same behavior? I have no way of testing this as I
have not moved up yet due to problems with CDR's and this very issue.
--
Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
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Re: [asterisk-users] asterisk registered as UA
Matthew Nicholson wrote: On Mon, 2009-02-09 at 23:25 +0200, Szasz Szabolcs wrote: Hi I registered my asterisk box to my SIP provider as an UA. For every call I receive on this trunk, I get the message That is not a valid conference number. I'm using Asterisk version 1.4.22, I had install the dahdi-linux and dahdi-tools and the conference is working between the phones registered to Asterisk PBX. What's wrong? Thanks. There is a problem some where in your configuration. Please post your sip.conf, extensions.conf, and meetme.conf files. You seem to be attempting to access a conference that is not configured (as the recording states). The most important thing is to have a context=context directive in the sip.conf entry for this connection so that asterisk knows where to route the calls when received, then in that context you either must have entries to the numbers callers from that connection might be dialing, or at least includes to other contexts that contain said numbers. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] AgentCallBackLogin no longer works after installing asterisk 1.6
The deprecation of Agent Callback login was announced in 1.4. Robert Broyles wrote: Check out this alternative: http://hostseries.com/agentcallbacklogin-alternative/ Regards, Robert Broyles oumar ndiaye wrote: Hi, My queue used to work fine until I upgraded to 1.6. I am getting the message: No application 'AgentCallBackLogin' for extension (default, 31001, 1) After some rearch I learnt that AgentCallBackLogin is removed in 1.6. Any one has a configuration that works in place of AgentCallBackLogin in 1.6. -- ond ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.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] AgentCallBackLogin no longer works after installing asterisk 1.6
Robert Broyles wrote: Check out this alternative: http://hostseries.com/agentcallbacklogin-alternative/ Regards, Robert Broyles I like what he came up ,with however it doesn't replace the agent callback login systems use of being able to make an agent press a key to accept a call, very important when people are logging in via cell phone and you don't their voice mail answering the call. In fact none of the replacements do that. FAIL -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Stopping chanspy followup
Jim Dickenson wrote: I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches chanprefix and spying will start. I would like it if there are no channels to spy on that the chanspy application exit. This can be done by changing line 673 of chanspy.c in the following way Old: if (res == -1 || ast_check_hangup(chan)) New: if (res == -1 || ast_check_hangup(chan) || !peer_chanspy_ds) Otherwise, as best I can tell, unless there is some error chanspy never exits unless the channel running the chanspy application hangs up, which I do not particularly want to do. In the interim I would recommend you make chat change and recompile. -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] Strange IAX2 registration issue
I have a single connection that seems to register ok but then becomes unregistered immediately. This is what I see with IAX debug turned on: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 6ms SCall: 1 DCall: 0 [76.25.248.23:4569] USERNAME: ashlawn-cfam REFRESH : 60 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGACK Timestamp: 2ms SCall: 2 DCall: 1 [76.25.248.23:4569] USERNAME: ashlawn-cfam DATE TIME : 2009-01-14 10:36:20 REFRESH : 60 APPARENT ADDRES : IPV4 204.144.134.114:1047 Here is what i have in iax.conf for this connection: [ashlawn-cfam] type=friend context= host=dynamic secret= disallow=all allow=gsm allow=ulaw The weird part is that port 1047. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.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] CDR Rewrite -- Questions to the users (Steve Murphy)
David fire wrote: 2009/1/12 Russell Brown russ...@lls.lls.com mailto:russ...@lls.lls.com Quoth Steve Murphy... Date: Mon, 12 Jan 2009 08:51:03 -0700 QUESTIONS: Which of the two would you see being useful to you? Obvious comment really but given LEG based CDR, one can determine the 'Simple' data but you can't work it the other way. I'd therefore find LEG based CDR more useful as the granularity (time on Hold, in Queue, Waiting on pre-xfer ring etc etc) would be good. -- Regards, Russell hi one question, i will need to rewrite all my apps that use the cdr? and the queue_log will be rewrited? thanks I have found that it is easier to use queue events rather than the queue log, I have my own custom build of asterisk in which I send everything that would be written to the queue log to the AMI, and then I use a perl script that watches for those events and writes them to a DB. I would recommend doing something similar as it makes adapting to changes in the structure easier. Anthony ___ -- 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] Upgrade to v.1.2.31 ... weird change
Tilghman Lesher wrote: On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote: I think it happened when I upgraded an install to 1.2.31 The variable CALLERIDNUM no longer works and CallerID(num) has to be used. I don't see why not. There has been no change whatsoever to that body of code. I know the initial one was being depreciated, but I didn't see any mention of it. I think you mean deprecated. Depreciation is an accounting term. The old variables for callerid where indeed put on the chapping block of deprecation, if you turn your cli verbosity to 3 or higher you should see warnings everytime the old variable is used. Anthony ___ -- 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] Security communication dilemma: your help needed
Kevin P. Fleming wrote: Tzafrir Cohen wrote: Suggested modification) X also signs the message with his public key. (If X doesn't want to, this automated procedure will not apply) I don't understand; if X signs the message using his public key, then recipients would need X's private key to verify the signature. Who would have that besides X? Generally an encrypted message is signed with the private key and decrypted with the public key, the point of public key encryption is not to hide the content of the message, but rather to insure that the content was not altered during transmission. Anthony ___ -- 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] Simple CDRs
Steve Murphy wrote: Sorry, I apologize for the 'uniqueID' field; I didn't invent it, or name it, and there is little definition for it. I think it's accidental that a transfer could yield two CDRs with the same uniqueID. I'm all for just simply dropping it. Maybe I will. I would ask that you do not lol. I use the uniqueid extensively to relate things like entries in the queue log to it's associated CDR. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.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] Simple CDRs
Tilghman Lesher wrote: On Friday 09 January 2009 01:14:37 Grey Man wrote: On Fri, Jan 9, 2009 at 6:37 AM, Tilghman Lesher wrote: I think Steve is as interested as anybody else in achieving a solution, but you're hand-waving when it comes to the establishment of a UUID. There is no such construct that we can use, but there are very deterministic methods (which Steve has enumerated) for producing the required uniqueness. And at the end of the day, what you need is the assurance that the algorithm used is indeed unique enough to produce no possible collisions. I've been hand-waving for over a year about the CDRs so you're right in that respect. There are no contructs in C for threads either but they are an abstraction heavily used by Asterisk, likewise linked lists etc etc. Correct, and WE'VE BUILT THOSE CONSTRUCTS. They are directly within the Asterisk code, and they may be viewed quite easily. At the end of the day whoever writes the code will implement it how best they see fit. I'm merely pointing out that there is already a very straight forward, standard way of generating unique ids that is used extensively and has a probability lower than 1 in 10^36 of generating a collision. In my opinion the methods discussed of incorporating a server name or timestamp into some kind of sequence to create a unique id are pretty fragile. That's not entirely true. For example, part of the algorithm for the Java UUID (I wasn't able to determine the entire algorithm) is to use the server machine's MAC address. That is part of a deterministic method of avoiding a collision. Note that the Asterisk Call Unique ID is itself very deterministic in avoiding collision within a single machine. It is impossible for a uniqueid, once generated, to conflict with another on the same machine, when referring to different calls. That's not to say that there can't be multiple CDRs with the same unique ID, of course, but they all refer to the same call. We are entirely interested in DETERMINISTIC methods of uniqueness, not random and hope-for-the-best. Given a truly random generator, it is possible for the same number to come up 100 times in sequence. That is part of what random means. It may be statistically unlikely, but it is just as likely as any other sequence. When it comes to fragility, using a random number for a UUID is NOT deterministic and MAY produce collisions. I may be over simplifying but I would have a serial number object that gets incremented anytime it is called and will be set to 0 at start-up. I would then use it to generate a UUID like this: MAC.serialid.64bit timedate Not only would this number be perfectly universally unique (as long as you dont falsify the MAC) but from a record standpoint it gives you easily parsable information in a single field, the id of the call for referential integrity, the machine that generated the uuid, the calls created since start at the time of the call creation, and the exact time of creation with microseconds. Just IMO. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Simple CDRs
Tilghman Lesher wrote: On Friday 09 January 2009 13:52:56 Anthony Francis wrote: Tilghman Lesher wrote: We are entirely interested in DETERMINISTIC methods of uniqueness, not random and hope-for-the-best. Given a truly random generator, it is possible for the same number to come up 100 times in sequence. That is part of what random means. It may be statistically unlikely, but it is just as likely as any other sequence. When it comes to fragility, using a random number for a UUID is NOT deterministic and MAY produce collisions. I may be over simplifying but I would have a serial number object that gets incremented anytime it is called and will be set to 0 at start-up. I would then use it to generate a UUID like this: MAC.serialid.64bit timedate Not only would this number be perfectly universally unique (as long as you dont falsify the MAC) but from a record standpoint it gives you easily parsable information in a single field, the id of the call for referential integrity, the machine that generated the uuid, the calls created since start at the time of the call creation, and the exact time of creation with microseconds. Of course, one of the problems comes in with: what do you do for machines which don't have a MAC address? We've been approached by individuals who use virtualized network addresses and don't have direct access to their MAC address (which is somewhat important for things like G729 licenses). What do you do for them? In cases of virtualization, at least in xen you can give a virtual machine access to a physical card, if not, then fake it using a fake mac address on each virtual machine in the research range of addresses, after all, you only really care about not conflicting with UUIDs in your own system. Anthony ___ -- 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] Simple CDRs
Dave Platt wrote: I may be over simplifying but I would have a serial number object that gets incremented anytime it is called and will be set to 0 at start-up. I would then use it to generate a UUID like this: MAC.serialid.64bit timedate I suggest reviewing RFC 4122, which discusses UUID formats in some detail. Your suggestion is very close to a standard version 1 UUID, which includes the host's MAC address, 60 bits of time information, and a 14-bit clock sequence value (which is set randomly at startup, and incremented if the system clock value is adjusted forwards or backwards or if the node ID changes). The time value has a 100-nanosecond resolution, which sets a lower limit to the amount of time which may be allowed to pass between UUID generation events. By my math this field won't wrap until after the year 5,000 C.E., so we have a while to prepare for the Y5237 wraparound problem :-) I had not reviewed that RFC, I am just a programmer and thought what would I do in this situation. So glad to hear my that my thinking wasn't that far off. Anthony ___ -- 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] Increase DTMF Tone Duration
Steve Underwood wrote: Wilton Helm wrote: The problem is simply the duration is too short (120ms), and the remote IVR seems to not detect them That sounds like an IVR issue. I've worked on some traditional PABXs and even designed some DTMF receivers. Any decent DTMF receiver should be able to reliably decode 80 ms tones, and a really good one can decode 40 ms. 120 ms should be a very generous duration. I shipped 80 ms duration to COs 20 years ago. Demanding 120ms is crazy. People just don't hold down the keys that long. You'd have horrible failure rates. Regards, Steve You can edit the duration of the tones in app_senddtmf.c and then rebuild asterisk. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.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] CDR Design
Steve Murphy wrote: Just to be pedantic, how would src_cid be different from the clid field that cdr's have now? and the same with src_exten vs. src; A simple example might help to let this sink into my brain. murf The main thing is that the originating number shouldn't be linked to the callerid. This way you can do things like allow no callerid while maintaining billing integrity. Here is the CDR columns for one one of my providers that exhibits this: *Field Number* *Field Name* *Description* *Type* *Length* *Example* 1 SwitchBatchNbr Sequential, positive integer assigned to each CDR file imported into the system Numeric Long Integer 5594 2 RecNbr Sequential, positive integer assigned to each CDR within a CDR file. Together with the SwitchBatchNbr, a unique combination. Numeric Long Integer 2354 3 SwitchNbr Unique number identifying the switch from which the CDR was processed or assigned Numeric Integer 13 4 CustNbr The unique, numeric number assigned to a customer Numeric Long Integer 1025 5 AuthCode The authorization code used in the call. Can be the Switch/Trunk combination (dedicated), ANI, Travel Card, 800 number, DID. Numeric Float 2145551212 6 AcctCd The Account Code dialed with the CDR Numeric Long Integer 2331 7 CallMMDD Call date at time of answer (MMDD format) Numeric Long Integer 20020131 8 CallHHMMSS Call time at time of answer (HHMMSS format) Numeric Long Integer 205618 9 DestNbr Destination Phone Number Char 18 2145551212 10 DialedNumber Dialed Number Char 18 12145551212 11 ThirdPartyNbr Third Party Number Char 18 2145551212 12 DestCity Destination city name Char 15 Dallas 13 DestState Destination state name Char 2 TX 14 DestOCN Destination OCN Char 4 9100 15 DestLata Destination LATA Numeric integer 552 16 IntraInter Flag indicating jurisdiction: 1=Intralata, 2=Intrastate, 3=Interstate, 4=Canada, 5=Intl, Mexico Numeric Integer 1 17 CallType Flag indicating type of call. See Appendix A: Call Type Codes. Char 3 OE 18 DurMinutes The rounded, billable duration of a rated call. Detailed to a tenth of a minute. Numeric Decimal 10,4 1.5000 19 CustRev The revenue computed for the CDR Numeric Decimal 10,4 0.0168 20 Surchrg The surcharge amount for the CDR Numeric Decimal 10,4 0. 21 OrigNbr Originating Phone Number Char 18 2145551212 22 OrigCity Originating City Char 15 DIR ASST 23 OrigState Originating State Char 2 TX 24 OrigOCN Originating OCN Char 4 9100 25 OrigLata Originating LATA Numeric Integer 552 26 SiteNbr Info digit assigned to CDR. Currently, Site Numbers: 7, 25, 27, 29, 70 are considered payphone Numeric Integer 0 27 SiteSurChrg Charge associated with payphone use as determined by the SiteNbr Numeric Decimal 10,4 . 28 ExtractSeqNbr Number used to designate a batch of CDR's that were extracted. If not used, value will be NULL. Numeric Integer 156 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED
Re: [asterisk-users] CDR Design
Steve Murphy wrote: Well, read my draft RFC, and see if I'm on the right track. Tune into CDR Design in the subject line in this email list, and let's toss this around and see if consensus is possible. murf First my apologies for this repost, my system date got messed up and this post looked like it was sent on Nov 5th :). One of the problems I generally have with cdr's in my multi-tenant hosted VOIP world, is that the src is inexorably tied to the callerid field, this makes it a pain when you have a billing system based on TDM billing systems that have not only a src field but an originating src field. This is what allows you to know what number placed the call but still allow things like no callerid. In a perfect world the fields would be src_cid src exten. When calls do not originate from within your dialplan (external to internal calls) most of these fields would be null or repetitive. I know many of you would say you know the originator of the call by the channel, but in a multitenant situation you can't have two sip devices named [100] so you use special ID's and have to do post-processing to determine that information. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] CDR Design
Atis Lezdins wrote: When i started to write this implementation, luckily i didn't had much expertise in telephony, so i did it from programmers point of view. There's even funny story about this in our company - we had some Project managers and Development managers hired later who had lots of experience in telephony, and at some point when discussing some minor problems with my implementation, they told me that this is not the way how to do it. Telco's do all processing at end of month, so this system won't last for long. Currenty everybody in our company probably would be very disappointed if they wouldn't be able to see fresh data in reports immediately. Regards, Atis Just because that is the way Telco's do it doesn't mean that it is the way it SHOULD be done, there is always room for improvement and fresh ideas. I also believe near realtime CDR is not only possible but should be used, the only thing I do once a month is long distance consolidation for billing, I use multiple LD carriers and all of their monthly records need to be normalized and consolidated with our records. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] CDR Design
Grey Man wrote: Another thing to be aware of as the wish list for the Asterisk CDR continues to grow is that right now Asterisk does not lend itself to accurately creating the most fundamental requirement of a CDR which is to accurately record at the very least the originator, destination, time and duration for EVERY call Asterisk processes. It's already proven to be a very hard requirement to meet for Asterisk given the original CDR design and implementation and to my mind there is no point trying to add more sohisticated behaviour - call flows, events, linked ids etc. - until it has been. The more convoluted any new design gets the less chance it has of ever getting implemented in the near future. Getting a basic accurate CDR system in place does not preclude future enhancements but without it they'll just add another few layers to the house of cards. Regards, Greyman. ___ -- 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 I agree with the fact that the base is broken and needs to be fixed first. -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] CDR Design
Steve Murphy wrote: Well, read my draft RFC, and see if I'm on the right track. Tune into CDR Design in the subject line in this email list, and let's toss this around and see if consensus is possible. murf One of the problems I generally have with cdr's in my multi-tenant hosted VOIP world, is that the src is inexorably tied to the callerid field, this makes it a pain when you have a billing system based on TDM billing systems that have not only a src field but an originating src field. This is what allows you to know what number placed the call but still allow things like no callerid. In a perfect world the fields would be src_cid src exten. When calls do not originate from within your dialplan (external to internal calls) most of these fields would be null or repetitive. I know many of you would say you know the originator of the call by the srcdevice, but in a multitenant situation you can't have two sip devices named [100] so you use special ID's and have to do post-processing to determine that information. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] CDR Design
We are suggesting the same thing, what you describe is multidimensional. If you think of the cdr's as being in a database, and say you wanted to have it show you all the calls today and all the calls that are associated with that call. Your select grabs the first dimension, a list of all calls. Then using the unique identifier of each call you build a second dimension of the related calls. [EMAIL PROTECTED] wrote: In order to avoid a multidimensional schema we could have 1 cdr per call leg. So , for instance, one call that had 3 different dial() commands as outgoing attempts would be described by 4 CDRs (1 for the incoming leg that has all the originating channel data and 3 for the outgoing legs that hold all the terminating channel's data). Those CDRs would be bound by a unique identifier field (the same for all 4). The terminating CDRs could be also identified by a increment field that indicates the order that the channels were called. Another issue is that failed attempts should also be logged because this is valuable info for many (or at least have the option to choose the desired behavior - which is available in asterisk as we speak). Anthony Francis wrote: It is my belief that before redesigning the CDR engine some time should be spent looking at current PSTN CDR formats and what information is kept in them. The main problem that I feel we face is that calls can be complicated, but we want the record of it to not be. In reality a CDR that incorporates all information about a call would have at least two dimensions. In the first you would have the base call record as we do now, in the second we would have the event list. The event list would be a time indexed list of event names and attributes, just as you would currently store event information. The event list would be your glue (with a bit of front end logic of course.) that would relate a call that dialed X external numbers to the X different new CDR's that generated. That would allow you all the call path info you could ever want. The most important thing would be a new config file that allows an administrator granular control over what information is important to them. And of course a keep it simple stupid mode that just writes the top level cdr as it does now. [EMAIL PROTECTED] wrote: I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things
Re: [asterisk-users] CDR Design
It is my belief that before redesigning the CDR engine some time should be spent looking at current PSTN CDR formats and what information is kept in them. The main problem that I feel we face is that calls can be complicated, but we want the record of it to not be. In reality a CDR that incorporates all information about a call would have at least two dimensions. In the first you would have the base call record as we do now, in the second we would have the event list. The event list would be a time indexed list of event names and attributes, just as you would currently store event information. The event list would be your glue (with a bit of front end logic of course.) that would relate a call that dialed X external numbers to the X different new CDR's that generated. That would allow you all the call path info you could ever want. The most important thing would be a new config file that allows an administrator granular control over what information is important to them. And of course a keep it simple stupid mode that just writes the top level cdr as it does now. [EMAIL PROTECTED] wrote: I think that the custom cdr back-end can be successfully used to maximize or minimize the CDRs detailing on a per-needs basis. Furthermore, the CDR() function gives plenty of room for even more detailing. In my opinion the detail level (fields) is not the issue with the CDRs generation nor is the lack of backends (asterisk gives a lot of different backends to store your CDRs). I find the issue with asterisk CDRs to be in the lack of proper CDRs generation for the B-leg of every call. If we want to really track what happens during a call through the CDRs one has to have all the details not only for the incoming channel but for the outgoing one as well. Furthermore, one needs to be able to tweak the B-leg CDRs like he does with the incoming legs. So what needs to be done in my opinion is record every B-leg CDR when such an event occurs and give the user access to the CDR info by extending the CDR() function (so that one can specify the channel of the CDR that is being tweaked) or creating a seperate one for the outgoing channels. Grey Man wrote: I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal (link ommitted to see if email gets accepted). After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong destination. Regards, Greyman. ___ -- 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
Re: [asterisk-users] hint priority with 50 channels
Loic Didelot wrote: Why do we put 80 character limits, computers have GB or memory? Loic Asterisk was written in c and they do have to declare how much memory should be reserved for a variable in c, so the programmer arbitrarily chose a number. They may have put some logic or investigation behind it, but thats pretty much it. If you don't like that chose, edit the definition in the source code and then recompile and voila! you have your longer string handling. ___ -- 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] hint priority with 50 channels
Just curious but why would you want to have a lot of devices all have the exact same state information? Philipp Kempgen wrote: Loic Didelot schrieb: I noticed that my hint priority stops working when I add to many extensions/channels. It looks like everything exceeding 80 characters is discarded. By stop working I mean the status is and stays Unavailable. This works exten = *1,hint,SIP/loicvoip1_1IAX2/loicvoip1_1SIP/loicvoip1_1_a1 This does not work: exten = *1,hint,SIP/bla1SIP/bla2SIP/bla3SIP/bla4SIP/bla9SIP/bla5SIP/bla6SIP/bla7SIP/loicvoip1_1IAX2/loicvoip1_1SIP/loicvoip1_1_a1 I tested on several asterisk 1.4 versions like 1.4.21*. Is this a bug or something like working as designed? It's by design. 80 characters is likely to be the limit. Is there another possibility to monitor a bigger number of channels? In Asterisk 1.6 you could build something with Custom hints and DEVICE_STATE(). Philipp Kempgen -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] How long will Asterisk 1.4.x supported/maintained
How do you go about determining this has happened? Tilghman Lesher wrote: Similarly, we will probably end-of-life 1.4 when a majority of users make the jump to 1.6. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] view the current calls and their codec
core show channels shows all channels and the first part of the ouput gives you the technology: *CLI core show channels Channel Location State Application(Data) SIP/xxx (None) Up Bridged Call(Zap/2-1) Zap/2-1 [EMAIL PROTECTED] Up Dial(SIP/xxx to get more output add the keyword verbose, to make it machine parse-able add the keyword concise. Tim Nelson wrote: 'oh323 show channels' I would assume... I don't have a box handy with h323 loaded to verify. Check http://astrecipes.net/index.php?n=89 Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - nik600 [EMAIL PROTECTED] wrote: And if i have an h323 configuration? Thanks On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels' assuming you want SIP... substitute sip for iax2 if you prefer... Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - nik600 [EMAIL PROTECTED] wrote: Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- 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 -- /*/ nik600 http://www.kumbe.it ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Using AMI to determine PRI Channels Used
It is easier than that, I have found that the link event watching is
buggy and not very reliable. Here is what I do in my PSTN context:
exten = _X.,1,Wait(1)
exten = _X.,2,NoOp(CIDName: ${CALLERIDNAME} - CIDNum ${CALLERIDNUM} )
exten = _X.,3,Set(GROUP()=ZAP)
exten = _X.,4,UserEvent(ZapCall|${GROUP_COUNT(ZAP)})
exten = _X.,5,Goto(dids,${EXTEN},1)
Then I do the same thing on the PSTN out part. ON my Perl event watcher
I have a callback for this event:
zapcall = {
'Event' = 'UserEventZapCall',
},
Which is handled by:
zapcall = sub {
my $event = $_[ARG0];
my $zc = $event-{content};
I would show you the rest but it doesn't matter, it just sticks the
number in a DB so we can do usage graphs and know when we need to
increase capacity. I also have it alert when all channels on a PSTN box
are used.
The benefit to using a group and a user event is that asterisk does all
the counting for you.
Richard Lyman wrote:
Godson Gera wrote:
On Mon, Nov 10, 2008 at 7:41 PM, David Budny [EMAIL PROTECTED]wrote:
What is the AMI command to see how many PRI channels are being used /
available?
Thanks
There is no direct command in AMI which will give you used channels number.
But you can easily keep track of the active zap channels, by catching the
respective events (events Link, Ringing,Hangup,Unlink etc etc ), on zap
channels, and storing them in a variable in your program.
When you receive a Connecting kind of event like Link,Ringing increase the
variable value, and when you receive Hangup or Unlink events decrease the
variable value.
Happy Hacking,
Godson Gera.
http://godson.in
Action: Status (this will show only active channels, parse out the
Zap ones)
and
Action: ZapShowChannels (this will list all Zaps, parse out what
you are looking for (PRI ones will have 'Signalling: PRI...'))
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Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
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Re: [asterisk-users] is it possible to deactivate RTCP?
So, you don't want any media? No audio, video, just sip packets? If you just want a sip router with no media look into SER. Klaus Darilion wrote: Hi! Is it possible to deactivate RTCP? (I am using 1.6) thanks klaus ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. David Gibbons wrote: I'm glad I'm not the only one who got that. I sent them a nasty response earlier this morning... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, November 06, 2008 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spam from DIDForSale [EMAIL PROTECTED] On Thu, 6 Nov 2008, Gordon Henderson wrote: didforsale.com have just sent me SPAM to the email address I use here. What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that I'll never used their services. Morons. The English have such a way with words :) I keep a local archive of the last 30 days list posts. Searching for didforsale.com shows: Buy unmetered VoIP DID from DidForSale.com is the signature for Jai Rangi [EMAIL PROTECTED]. A wolf in the sheep's pen? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Spam from DIDForSale [EMAIL PROTECTED]
http://en.wikipedia.org/wiki/Jacque_Fresco A resource based economy. Greg Woods wrote: On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: Gotta love this list being farmed for spammers now. I am sure they call it targeted delivery or some such nonsense. I can't wait for capitalism to completely fail, then there won't be any spam. Socialism has already completely failed. What should we do, go back to a barter economy? :-) ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] SIP # DTMF
On many phones # sends the call. Rodolfo Alcazar Portillo wrote: Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only 333 What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] PSTN Simulator
Another asterisk box set up to be the network side of that link? mark morreny wrote: Hi, I have Asterisk setup to run on SS7, and I would like to test it out before getting the line from my telco. Is there any testing or simulation tool that I can buy to simulate a E1/SS7 link? Could anyone give some suggestions? Thanks alot for your help in advance. Regards, Mark ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DNS Query Overload
Adam Lovegrove wrote: Hi, Thanks for the suggestions. I've had a look at nsswitch.conf, it's set as: hosts: files dns. I'm guessing this is correct. I haven't tried the caching DNS server yet, but I really don't understand why Asterisk is doing dns lookups similar to2019-b6912730.mydomain.com http://2019-b6912730.mydomain.com for every call. The part after the - seems to be unique for every call, so why would there be a DNS record? I'm confused. Most distros come with a caching daemon. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [FreeBSD 6.3] Right-way to recover Zaptel?
If Asterisk is running that will happen. Make sure to shutdown asterisk cleanly before doing that. Anthony Vincent wrote: Hello I'm running Asterisk 1.4.20.1 on a FreeBSD that I compiled from the Ports collection. It's the second time I'm having an issue with a FXO card and/or the Zaptel driver. I couldn't figure out what else to do, so I just rebooted the server, but I'd like to know what happened, and whether there's a less drastic solution. Here's some infos: === # /usr/local/etc/rc.d/zaptel stop zaptelkldunload: can't find file wcte12xp.ko: No such file or directory kldunload: can't find file wcte11xp.ko: No such file or directory kldunload: can't find file wct4xxp.ko: No such file or directory kldunload: can't find file wct1xxp.ko: No such file or directory kldunload: can't unload file: Device busy kldunload: can't find file wcfxo.ko: No such file or directory kldunload: can't find file tau32pci.ko: No such file or directory kldunload: can't find file qozap.ko: No such file or directory kldunload: can't unload file: Device busy Sep 6 19:11:12 freebsd kernel: kldunload: attempt to unload file that was loaded by the kernel # kldstat Id Refs AddressSize Name 19 0xc040 7a05b0 kernel 21 0xc0ba1000 5c304acpi.ko 121 0xc2d6c000 19000linux.ko 131 0xc3ba9000 32000zaptel.ko 171 0xc3c0d000 a000 wcfxs.ko # kldunload -i 13 kldunload: can't unload file: Device busy # kldunload -i 17 kldunload: can't unload file: Device busy === Thanks for any tip. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination
Shaun Wingrin wrote: The setup is as follows: SIP phone registers via international link to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 via Zaptel Channels need to be hairpinned from Box 1 to 2. How is sip.conf configured on Box 1 and 2 so that we don't get an error: Failed to authenticate user when 1's extensions.conf uses SIP to dial Asterisk Box 2 . How do we ensure that RTP traffic flows from SIP phone registering at 1 directly to 2 without first passing through 2? Tx Shaun ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This happens through a sip re-invite, the problem you seem to be having is that box 1 is not authenticated to send calls to box 2. Anthony /Everything should be as simple as possible, but no simpler - Albert Einstien/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP vs. SIP-DECT
Lets not forget that the DECT specification does allow for data transmission. THere is no reason that in the future you would not be able to have integrated services over DECT. Michael Graves wrote: On Fri, 29 Aug 2008 09:58:56 -0500, Karl Fife wrote: Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer that was engineered from the ground-up with the design priorities of a wireless endpoit. I notice that the standby times of Wi-SIP vs. SIP-DECT are a great illustration of this point. I guess there's no low-power way to participate in a WiFi network, hense standby battery life that sucks in Wi-SIP. I've never actually demoed a Wi-SIP phone on premesis, but if the range of my WiFi LAPTOP vs. my DECT 6.0 headset is any indication, (DECT more than double the range) I'd guess it to be quite hard to make a case for Wi-SIP unless you're doing some straight-up network application integration right onto the phone. Can anyone speak to this? I've used both fairly extensively in a home office setting. DECT is the clear winner. That said, the current crop of wifi APs and SIP handsets can do a good job, but it's gonna be more work and maybe a little more expensive that you think. You need newer APs with WMM. Unless there's a truly compelling reason to go with converged voice+data over wifi I'd recommend DECT in most cases. Michael -- Michael Graves mgravesatmstvp.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FTC Bans Prerecorded Telemarketing Drivel
Jay R. Ashworth wrote: On Thu, Aug 21, 2008 at 02:15:58AM -0400, Alex Balashov wrote: I would be curious to know where, in this classification, fall various telemarketing schemes that are technically not cold-calls, but are generated from leads that come from customer-provided information, but where the customer does not know explicitly that they are signing up to receive calls. For instance, this is common in a number of industries such as financial services. You do a search to get a quote on something, and provide your phone number in the process, although the phone number bears no relation to the submission and is just an ancillary required item. Several places' telemarketing organisations call you back in response. For example, lendingtree.com. Is this a solicited call? In order to classify that as a solicited call, I believe, you have to have language *on the form the customer fills out* that says they're authorizing you to call, and you have to be able to produce ink-on-paper if the FTC ever calls you on it. IANAL. YMMV. Cheers, -- jra Actually in the US all you have to do is provide some proof of a business relationship with them. Companes get away with calling you if you have ever bought even one item from them. -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callfiles/manager api originate call fails
Rizwan Hisham wrote: Hi all, asterisk is giving me tough time. its been 3 days I am trying to originate outgoing call using manager api/callfiles. I would say remove the @TRUNK-OUT part and make sure that the context you send the call to knows about sending calls to the outside world. -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help...i cant do more...
David Thomas wrote: On Fri, Apr 25, 2008 at 4:38 AM, Bruno Pereira [EMAIL PROTECTED] wrote: Thanks for the answers. I need to say that this command is executed from another machine, with the command ssh because in ocalhost is all ok, with sudo or with root. I will try that trace to see if it helps me, but the bg probem is start the service from another machine with ssh . Did anyone ever find a solution to this issue. I have the same problem when trying to start asterisk from another computer via SSH. It starts fine on the local box, but over SSH it just hangs forever. I am using root as the user, and issuing the command: ssh 10.0.0.10 '/etc/init.d/asterisk start'. Thanks! Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The real solution is to run Asterisk as a service, but if you must have it run from a console then I would suggest starting it in a screen. That is, make sure you have screen installed, run it, and then start asterisk. After that disconnect from the screen session by pressing ctrl+a and then d. To reconnect to the screen session at anytime you simply do screen -r. The issue with simply running the asterisk command from an ssh session is that its process is started as a child of your remote shell. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT: ServerBeach for VoIP
My organization may be able to help you out on this, I am forwarding this email to my sales team. Kristian Kielhofner wrote: Hello, I'm looking at getting a dedicated server from ServerBeach to host some light Asterisk/VoIP/SIP stuff. Has anyone used them for this before? I'm pretty sure I've heard good things (in general) about them but VoIP is a very different animal than web hosting - especially for the network (obviously). ServerBeach uses the Peer1 network which looks pretty good. In fact that's how I found out about them in the first place. Or maybe I can do better than ServerBeach? Does anyone know of a dedicated hosting provider that meets the following specs: - Multiple physical datacenters available by request - Well peered network with multiple Tier 1's (Level3, ATT, Qwest, Verizon Biz, etc) - Dedicated servers running Linux (preferably CentOS) Ideally I'd like to be at $150/mo or less. Bandwidth/peering is important but transfer isn't really an issue - SIP/RTP is just a bunch of small packets! :) Other hardware specs don't matter much either. I'd rather have a Pentium 2 running on an awesome network than have an Athlon 5000 with nothing but dirty bandwidth. Any ideas? -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Penalties not working properly
Syed Nasruddin wrote: Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr To do this the sip device must return a sip busy message from the device already on a call from the queue. Make sure that you disable call waiting on this line appearance. Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
Your using a Linksys right? you can use the fxo port and send DTMF. Chris Bagnall wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] next priority from Dial in Asterisk 1.6
It is reading [EMAIL PROTECTED],,tTwWg as the device string.if you are
dialing to a sip connection called ip you would say
Dial(SIP/IP/${EXTEN},opts)
Carles Pina i Estany wrote:
Hi,
On Jul/23/2008, Carles Pina i Estany wrote:
I'm testing Asterisk 1.6 (from SVN). In my dialplan I have:
--
exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg)
exten = _00X.,2,Verbose(After Dial)
--
Also this doesn't work either:
exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg)
exten = _00X.,n,Verbose(After Dial)
I mean, like before, after some SIP responses like 484 is not executing
the after dialing command.
In Asterisk 1.4.21.1 it was working as I expected.
Is it a feature in Asterisk 1.6? or a bug?
After 404 it's going to next priority, but not after 484.
Thanks,
--
Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
___
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Re: [asterisk-users] Zap Channel Oddity
Not dialing a 1? Make sure it as actually send the call out the zap interface. Whats the console say? :) Jeremy Mann wrote: Can anyone help me start to diagnose why a Sangoma A200 wouldn’t dial out LD? Local calls are fine, incoming is fine, just no LD. Bell tech has been on site and plugged into lines with his test set and was able to dial LD just fine, so it’s not a LEC issue. No errors in asterisk console, using zaptel 1.4.11 and sangoma drivers 3.2.6, asterisk 1.4.18 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to monitor Asterisk logs ?
perl script. Olivier wrote: Hi, How can I be notified anytime a given warning message appears in Asterisk logs ? I've got a running system that produces cause 34 warnings (Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)) once or twice a week. I would like to like to be notified (by email, phone, ...) anytime such warning message occurs in log file. I was thinking of using logwatch but wondered if anything better exists. Any advice ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Good light codecs like speex, and minimal feature sets. C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Steve Underwood wrote: C. Savinovich wrote: I am puzzled by the quality of magicjack. I keep trying to figure out how they can the quality be that adequate. Since Skype also has an excellent quality, that leaves me to believe that software based calls (softphones) could have and advantage over hardphones, provided there is a parameter that those 2 companies are addressing. Anyone's thoughts on this? CS I don't know what Magic-jack does (I've never actually seen one), but I know the key thing about Skype that impresses people - its wideband voice codec. A lot of people poo-poo the idea that wideband voice has value in a phone call. They are either close to deaf, or have never tried it. Clarity is profoundly improved. Skype seems to use various tricks to keep the packet flow smooth, but its wideband that makes it sound better than the PSTN. You might think a standard phone plugged into an adaptor, like a Magic-jack, would be limited to narrow band voice, as that is all the phone was designed for. It turns out most phones only aggressively filter at the low end of the band. They let a lot of energy above 4kHz through, and they do generally sound better through a wideband codec. Many modern line interface chips are actually capable of running in a 16k samples/second mode, even though most are programmed for 8k samples/second. I think the ones on the TDM400P type cards can. Some from Silicon Labs certainly can, and chips from Zarlink and others can. If Magic-jack sounds impressively clear, a wideband codec would be my guess. Regards, Steve Like I said, Speex. It features Narrowband (8 kHz), wideband (16 kHz), and ultra-wideband (32 kHz) compression in the same bitstream. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Disconnect very often
Sounds like a network connectivity issue. Start at your physical layer and work up. Tariq .. wrote: Greetings.. i'm having a disconnection problems with Calls comming to my Call Center.. i'm using the free version of G729 .. and i'm starting to suspect it would be the reason.. i just need to know if it's possible or there will be other problems?? i asked the technician in the location to do a ping to our DNS and there seems to be a problem.. one of them doesn't response and the other one has lots of timeouts.. he noticed that the disconnection of the call occures at the same time a timeout happens to the DNS .. so my question will be the following.. is it a CODEC problem? G729 more sensitive to internet than G723? Regards Tark Sawah -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error: conflicting types for ‘bool’
Robert McNaught wrote: Hi All, Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos 5, and getting the following error trail on make. Googling the issue has found one user who tried: seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Roman The only thing non-standard with the machine is a RAID controller which was installed from a floppy in the first part of installing linux linux dd from the anaconda prompt. I have since done a yum -y update kernel kernel-devel and rebooted the machine. The running kernel is the same as the sources Linux localhost.localdomain 2.6.18-92.1.1.el5 #1 SMP Thu May 22 09:01:47 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Does anyone have any troubleshooting advice on this? Thanks in advance Robert .o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ modules make[2]: Entering directory `/usr/src/kernels/2.6.18-92.1.1.el5-x86_64' CC [M] /usr/src/zaptel-1.4.11/kernel/wctdm24xxp/../voicebus.o LD [M] /usr/src/zaptel-1.4.11/kernel/wctdm24xxp/wctdm24xxp.o CC [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/../voicebus.o LD [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/wcte12xp.o CC [M] /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /usr/src/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /usr/src/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here make[4]: *** [/usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 make[3]: *** [/usr/src/zaptel-1.4.11/kernel/xpp] Error 2 make[2]: *** [_module_/usr/src/zaptel-1.4.11/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.1.el5-x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.11' make: *** [all] Error 2 ___ -- 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 Might be the version of the gnu compiler you have, make sure you do a yum groupinstall Development Tools, and then make sure you build libpri first. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
Eric Wieling wrote: Doug Lytle wrote: Eric Wieling wrote: Remove the qualify= option from sip.conf. Also make sure the DISABLE CDP in the Polycom's boot menu. That didn't help and CDP is off by default, the phones still couldn't receive/send calls when in this state. I've sent an employee out to grab a replacement NIC. Hopefully this will fix things. Based on the SIP poke message you pasted in an earlier message, the qualify= option you used is virtually guaranteed to cause SIP poke problems. Did you ever try turning off all phones, flushing the lease table and bringing the phones back up? ___ -- 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] Concept Clarifications
I would suggest actually looking at the arguments for the queue command. There is an option to play a ring instead of hold music. To look at the syntax for any command in asterisk, from the cli type show application command name. So for you: show application queue -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP Joseph L. Casale wrote: Sounds like you want a ring-all queue. Appreciate that pointer. So if I make a queue with a strategy = ringall and add all the extensions I want in it, then send the incoming calls from the sip did into it, do the callers experience being put on hold, or do they experience a ringing line until it is answered? Thanks! jlc ___ -- 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] One way sound when Using Dial cmd without t option (SOLVED) Need explanation
The t, much like reinvite = no keeps asterisk listening to the audio stream to detect dtmf input if dtmf mode is in-band, what is happening is that the sip reinvite is failing, due to a firewall rule or a routing problem and you end up with only one connected RTP stream. Asterisk does not require the t option. Anthony Moe Navid wrote: Thanks Tony for you reply. Do you have any idea why Asterisk require t in Dial command? Cheers, Moe On Sun, May 18, 2008 at 1:14 AM, Tony Mountifield [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED] mailto:[EMAIL PROTECTED], Mohammad A. Navid [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm implementing a simple calling card feature for testing purpose. I have a DID number, when I called my DID number and enter the phone number to call, Asterisk would dial the number for me but the sound was only one way. After hours of struggling with the problem, I found out that I need to add t to my dial options, this is the correct way of dialing out: - Dial(SIP/carrier/310555|20|t) Now I need to know what was going on? Why with option t both parties can hear each other, but without option t in dial cmd only one party could hear? Another interesting issue is, if I use Answer() command at the begining the sound becomes one way even if I use t in options. Try adding reinvite=no to the sip.conf or users.conf definition for your SIP service provider. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] where did the switch statement come from?
The Switch statement is used to bring any external dialplan configuration into an asterisk context, such as realtime configs and even DUNDI. http://www.voip-info.org/wiki-DUNDi To answer your actual question. Rizwan Hisham wrote: where can i find details about both switch statement and dundi. On Mon, May 19, 2008 at 4:37 PM, Alexander Lopez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: The switch statement allows you to 'include' a context from another machine into your machine. Problems with it was if the other machine was unavailable, or even slow to respond, your machine would hang until it timed out. DUNDI has since replaced the functionality of the switch statement and given you so much more in return *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Rizwan Hisham *Sent:* Monday, May 19, 2008 6:59 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] where did the switch statement come from? Hi all, I have looked up the applications and function in asterisk but i could not find the help for the switch statement which is used in several places in sample extensions.conf file. i am using asterisk 1.4.2. http://1.4.2. On voip-info.org http://voip-info.org the switch statement seems to be used to connect 2 asterisk servers, but i could not find a satisfactory explanation for the this statement. Can anybody help me understand the switch statement? -- Best Regards Rizwan Hisham ___ -- 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 -- Best Regards Rizwan Hisham ___ -- 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] DHCP Failure screws up system
Watch your router and see if the arp entries are changing for the IP's as this happens, I am very willing to bet you have IP address conflicts on a mass scale and should consider shutting down all of these phones and bringing them back up so they all get leases out of the new servers pool instead of being a miox between that and your old one. Doug Lytle wrote: Maybe someone could point in the right direction. I have a small facility that's running around 40 Polycom 301/501 phones, Asterisk 1.4.18 running under Mandriva 2007.1. The phones were assigned a DHCP address in the 10.10.10.x range. Today, the DHCP server failed and to get them back online, I loaded the dhcp-server onto another system (Also running Mandriva) and copied the dhcpd.conf to that pc. Now, after bringing that system online, the phones are bouncing up and down every few seconds with the following errors: [May 19 14:32:23] NOTICE[3231]: chan_sip.c:15778 sip_poke_noanswer: Peer '4231' is now UNREACHABLE! Last qualify: 39 [May 19 14:32:50] NOTICE[3231]: chan_sip.c:15778 sip_poke_noanswer: Peer '4276' is now UNREACHABLE! Last qualify: 40 [May 19 14:33:02] NOTICE[3231]: chan_sip.c:15778 sip_poke_noanswer: Peer '4247' is now UNREACHABLE! Last qualify: 38 I've got there qualify=500 A few seconds later they become reachable and then we start all over again. When in the unreachable state (But I can still ping them from the phone system), they can't access the phone system). Little info on the dhcp. It has two interfaces, eth0 is for known hosts, eth1 is for unknown hosts. The phones are assinged as unknown hosts. The Asterisk system has two IP addresses attached to the eth0 interface. eth0:FWB1 is on the 10.10.10.15 where the phones register. I've rebooted all the phones, Cisco switches and even the phone system. Makes no difference. Any suggestions would be appreciated. ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Concept Clarifications
Joseph L. Casale wrote:
Hi,
My system is setup and working, I can dial out, in and to demo extensions
that play music etc.
I would like to read up on some final topics before I get it running in
production but don't really
know what to look for.
If an incoming call from a SIP DID is to ring across {n} phones for example,
what is this procedure called?
How dose one write this up elegantly so if a new phone is purchased, it can
be added to a group and
be included in all pre-existing config?
Thanks!
jlc
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Sounds like you want a ring-all queue.
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Re: [asterisk-users] Polycom XML Files / asterisk
I am confused how TFTP is less secure than HTTP. TFTP does not allow any browsing, ever. Neither technologies will allow the device to authenticate before downloading a configuration file, and both are easily secured by only permitting connections from specific hosts. Robert McNaught wrote: Yes, perhaps a script would always be better than hand-touching these files, and getting an XML editor only really makes it easier on the eyes. On the same subject, I have noticed that Snom and Linksys phones do not support FTP provisioning - only TFTP and HTTP. With TFTP being an insecure option for a hosted architecture, is everyone moving to provision Polycoms with HTTP, so that both can be auto-provisioned via Option 66. One thing I found is that, with option 66 in a LAN router, you cannot specify more than one protocol. Has anyone had any problems provisioning Polycoms with HTTP? On Thu, May 15, 2008 at 1:35 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Robert McNaught schrieb: Does anyone know how to apply a style sheet to the polycom automatic provisioning XML files? Why should applying a stylesheet be different than for any other XML files? Even better, does anyone know of a web-based XML editor where you can just edit the files from a browser directly ie entering in phone number, display name, proxy address etc. From what I gather, most people are just using Notepad to change the files then upload them, or vi from the command line, which is fiddly and time-consuming. Just use your preferred editor. Nobody forces Notepad or vi upon you. Even better: Generate the config files with Perl/PHP/insert favorite language. Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, 26.-27. Mai - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] How to handle multiple IPs from one SIP carrier
[EMAIL PROTECTED] wrote: On my SIP carrier, I register to a proxy sipconnect.dal0.cbeyond.net which ends up being 192.168.22.212 (They supply a T1 bundle) #sip show peers Name/username HostDyn Nat ACL Port Status snip Generic-8174691929/817469 192.168.22.212 N 5060 OK (41 ms) Yesterday, they had a problem with their primary server and reverted to a backup server for about 5 minutes. As chance would have it, I received a call to one of my DIDs just before and just after the switch. As you can see below, the first call was on their primary server and the Found peer finds the Generic-8174691929 peer I have set up. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.22.212 : 5060 (NAT) Found peer 'Generic-8174691929' Found RTP audio format 0 Found RTP audio format 100 However, just after they changed to the backup service, I received the call below. Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.25.212 : 5060 (NAT) Found no matching peer or user for '192.168.25.212:5060' Found RTP audio format 0 Found RTP audio format 100 Since it was a different IP address, it found no matching peer and failed to find a valid context to send the call to. How should this be addressed in Asterisk to allow for such an incident? Bill ___ -- 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 This is why Asterisk recommends dual registration. You reg with them for out and the reg with you for in. :) ___ -- 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] Help Please - Asterisk MYSQL interface seems to be eating data
Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] StatusComplete is getting me sick !!
Did you recently upgrade? If so, from what version to what version? Nestor A. Diaz wrote: Hello Asterisk People. Asterisk have a really annoying bug, i use frequently the manager status command and when asterisk decide not to show the statuscomplete event, it really don't show the statuscomplete string, in fact none of the AgentsComplete, QueuesComplete' are shown I use it for monitoring a queue, but this is really getting me sick. Does anybody have to deal with this issue and found a solution ? something that doesn't rely on restarting the asterisk server, since this is not a viable mechanism slds. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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 using 100% of CPU
Plus that originate is going to call the sip device, and upon answer connect it to extension 0 in the internal context, is that what you wanted? Tilghman Lesher wrote: On Friday 25 April 2008 15:23:05 Chris Elliott wrote: If I reverse the situation it gets a little better. Asterisk doesn't use 100% of the CPU, but until SIP/exten-20 answers, the manager interface doesn't respond. So I can't hangup the line using the manager API if SIP/exten-20 doesn't answer. SIP/exten-20 is a SPA3102 FXS. Here is that example: Action: Originate Channel: SIP/exten-20 Context: internal Extension: 0 Priority: 1 The reason Manager doesn't respond is that it's waiting for a result code to give you. If you don't care, use the Async: yes option to the Originate action to get AMI to continue past that point. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Next step in extensions.conf after answer the phone in Queue
Atis Lezdins wrote: Atis Lezdins wrote: Queue will continue if called person hangs up (and there's no option). If caller hangs up, call goes to h extension in same context. Just the same way as Dial with 'g'. There's a change in 1.6 that allows called channel to continue if caller hangs up, so probably something like this could be applied also to Queue (or was that actually working with using Local channels?). On Wed, Apr 23, 2008 at 8:18 PM, Al Baker [EMAIL PROTECTED] wrote: Why would you want a channel to continue after the caller has hung up. I clearly am missing something here because I can't see what good that would be. What do people do with this Continued Channel ? What is is used for ? How Does having it help you ? ??? To play something to called party. I'm not familiar with that feature too deep, but I guess it's not caller channel but called channel that's continued. Regards, Atis I am guessing something to the tune of missed a call from number press 1 to call them back now.. That is a good feature idea. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] G723 pass thru
More importantly, for it to pass-through you need something that processes g723 on the other end. If Asterisk is terminating the call by handing it off to the PSTN or to another phone that does not do g723 then Asterisk must transcode and that requires the license. Eric Wieling wrote: allow=g723.1 or allow=g723 (I don't remember which). aby azid wrote: Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] sometimes UNREACHABLE!/REACHABLE doesn't appear in the log when it should
The asterisk code is full of fun things where it checks for things like that in multiple places but doesn't always handle every instance of the same check in the same way. This is getting resolved piecemeal and will eventually be minimized as the application develops, but I do not think things like this will ever completely go away. fadey wrote: Hi, everyone. I'm having a problem with qualify=yes sip.conf option. Sometimes, when a device registered with asterisk goes offline, I'm not getting a message about it in /var/log/asterisk/messages log. Sometimes the same happens with REACHABLE message, when a device comes back online. I'm pretty sure asterisk is aware of a device's current state. I can see it with sip show peers command. Is this a bug? If not, how could I achieve the desired behavior: every time a device changes its state, a message in the log appears about it. Thanks in advance and sorry for my English. I'm still learning :-) ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Chanspy on Asterisk 1.4.19
I saw a patch attached to that bug report, just download it run patch
and then make clean make install, restart asterisk and you should be
smokin.
Mike wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Michelson
Sent: Thursday, April 17, 2008 17:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Mike wrote:
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly).
The other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Steve
Rawlings
Sent: Thursday, April 17, 2008 14:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Guys,
Sean Bright wrote:
Steve Totaro wrote:
Should one have to change their dialplan for functionality
to remain
the same in the same version?
I wasn't suggesting it wasn't a regression, just making the
OP aware
that he can pass multiple arguments to a dialplan
application (i.e.
ChanSpy(|bg(2000)))
He mentioned that he was able to get it to work in 1.4.19
by passing
the bridge argument ('b') but didn't seem to be aware
that he could
also pass his original argument list ('g(2000)') as well. Seems
easier to just work around the problem with the
additional argument
than to backport the application.
Yes I was aware of multiple arguments, I did try
chanspy(|bg(2000)),
I tried all combinations I could think of. Although maybe what I
should have said was I tried
chanspy(|b) just to prove chanspy itself was working at
all (and it
was), with chanspy(|bg(2000)) the 'spygroup'
element didn't work, it just spied on every active call.
Anyway, I've raised a bug report as requested by Jared at Digium.
Steve
This was an incredibly subtle bug that was introduced into
1.4.19 when the other work was done on chanspy to fix crashes
and deadlocks. It has been fixed in 1.4 in SVN revision 114226.
Basically, chanspy was a crapshoot if you didn't specify a
first argument, because the function intended to walk through
the list of active channels would always end up returning the
first channel it found. If that happened to be a spy-able
channel, then great, otherwise you'd never spy on anything.
Mark Michelson
Mark,
I added a first argument. Here is my line now:
exten = *012,n,Chanspy(SIP,qg(GROUP_NAME))
Unfortunately, that still crashes Asterisk once out of 3-5 times. Is there
anyway to absolutely prevent crashes with this bug in vanilla 1.4.19?
Thanks,
Mike
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--
Thank you and have any kind of day you want,
Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
[EMAIL PROTECTED]
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Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Tzafrir Cohen wrote: On Thu, Apr 10, 2008 at 01:08:21PM -0400, Doug Lytle wrote: Joshua Kinard wrote: send core set verbose 999 when connecting in. I'm running Mandriva and found the line that had -vvv. I like mine at 15, so just put 15 v's on that line. Worked great. 15? What do you need that for? IIRC the highest verbosity level is 5. anything more than that doesn't change the clogging of your logs. This one goes to 11 -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- 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] Jumped from 1.2.7 to 1.4.19, missing CLI colors
Tzafrir Cohen wrote: On Wed, Apr 09, 2008 at 08:00:38PM -0400, Mike wrote: Ah, not bad. When I start asterisk with /usr/sbin/asterisk -c I get the colors, but if I start it without -c and then connect to the console using /usr/sbin/asterisk -r I get no color. Since I want this to be running in the background, how do I fix this so I get to have my cake and eat it too? The patch is rather trivial. Just make Asterisk pretend that it is vt100 (or whatever) if it is running as a service. I cant get color using asterisk -r on 1.2.17 or 18 either. ___ -- 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] Star Wars Echo Sound
If the problem is specific to certian inspections I would verify the LAN segments involved in connecting those devices. Rob Schall wrote: Actually, its just the opposite... The call is okay for a few seconds, then the odd echo kicks in. When the training isn't turned on, it takes 20 seconds to so to kick the echo. With the training on, it works great except for this bug. Several of the people using the same * system but different phone stations are not seeing this problem. I saw someone else believed it was a softphone issue. Is it possible that its not a sangoma problem, but rather a polycom 501 issue? I just want to start putting the grind to the correct people. Rob Chris Earle wrote: I wanna say that's the echotraining taking effect. What it does is try to cause some echo so it can dynamically reconfigure the levels on the fly -- right at the start of the call. I know this happens with digium cards -- not sure if the Sangoma cards behave the exact same way. It's only at the start of the call right? once that occurs, the EC is kicked in and everything is fine? -- Chris Earle System Solutions Specialist, Network Technologies Division CBL Data Recovery w: http://www.cbltech.com Rob Schall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars type of sound. I expierenced this myself when talking to them. By this, I mean you hear a few words from them, then a few seconds lagging behind, you'll hear a muffled (darth vader) version of the same thing. Has anyone seen this? Thanks, Rob ___ -- 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 -- Thank you and have any kind of day you want, Anthony Francis ___ -- 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] load balancing SIP extensions
Vieri wrote: What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there I tried to use DUNDi on my local servers but I can't seem to make it work. Most howtos out there explain the use of DUNDi when the extension ranges do not overlap. So in my case where both *1 and *2 have the same local extension range 4XXX, can I go the DUNDi route or should I stop bashing my head on that and explore another solution? If someone has configured a similar system then I'd greatly appreciate some tips. I read a few dundi docs like http://www.voip-info.org/wiki-DUNDi. Thanks Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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 Have you tried placing the sip registrations in a db using realtime? ___ -- 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] Best ATA. Period.
SIP wrote: Adam Moffett wrote: In all seriousness, my requirements were a little silly. A Cisco router can fail just as a netgear router can. But I think we would find Cisco failures to be statistically less likely. I also think we can agree that not all devices of a certain type are created equal. Do you have any opinions on which VoIP products are more likely to be consistent and reliable? Realistically, I've had issues with every ATA I've used to SOME degree. The Leadtek BVA series has numerous issues. I've had bizarre things occur in all of my Linksys/Sipura adapters(2000,3000,3201) (issues with timeouts on a lost connection, NAT traversal, etc). My Grandstream HT486 and 488s have intermittent dialing failures. I've had a lot of issues with the Audiocodes MPs. The only ATA I've NOT actually had any issues with has been my Grandstream HT386. Granted, I have issues with its capabilities overall, but on the whole, it's the only one that's not simply had some weird random failure as the others have. Does this mean I'd recommend an HT386 as solid testing piece? Heavens no. I'd probably recommend the Linksys SPA3102. But be aware that there ARE issues with just about all of them, and I doubt there have even been enough variants sold/used by everyone to merit statistical analyses. What you're going to get for recommendations will be, at best, anecdotal. N. I have had good luck with Cisco's ATA's. ___ -- 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] Automatically start after restart
I actually use daemon tools http://cr.yp.to/daemontools/daemontools-0.76.tar.gz I like it because its log handling features, it takes the stdout of asterisk and puts it in a log directory and automatically rotates the files. Doug Lytle wrote: bilal ghayyad wrote: Any script or something that can do that? The scripts are located in the Asterisk source directory under contrib/init.d Doug -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] restart asterisk daily
Al lists wrote: Always rely on free -m to see how much free memory you have not top. in terms of memory leak, i have asterisk running on servers with uptime of 400 days (CentOs), if there was any leak, i'm guessing i would have crashed server long time ago. On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you want to flush your disk cache to see how much memory is being eaten cache pages, try this: echo 3 /proc/sys/vm/drop_caches - ast erisk [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: ___ -- 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 You will see asterisk behave its worst with multiple queues and heavy dialplan logic. I restart my boxes with queues everynight at midnight just to reset the queue stats displayed with show queue. ___ -- 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] How to balance traffic between 2 gateways ?
Atis Lezdins wrote:
On 2/7/08, Olivier [EMAIL PROTECTED] wrote:
Hi,
Is it possible and safe to split or balance outgoing calls to 2 different
sip-to-tdm voice gateways ?
I need 5 E1 ports and the boxes have 4 ports each.
Setup would be :
PSTN --1xE1-- Gateway1 ---2xE1 PBX TDM phones
|
LAN -- Asterisk - SIP
Phones
|
PSTN --1xE1-- Gateway2 ---1xE1 PBX TDM Phones
Regards
Sure:
context dial-out {
_X. = {
if (${GROUP_COUNT(gw2)}${GROUP_COUNT(gw1)}) {
Set(OUTBOUND_GROUP=gw1)
Dial(SIP/[EMAIL PROTECTED])
} else {
Set(OUTBOUND_GROUP=gw2)
Dial(SIP/[EMAIL PROTECTED])
}
}
}
Regards,
Atis
Old variable syntax but this flips which gateway is used with every
call, and fails over to the other line in the event of any negative
status return, of course it could be cleaner, if both gateways where
down this would make an infinite loop.
[macro-lb]
; ${ARG1} - PhoneNumber
exten = s,1,GotoIf($[${LOADBALANCE} = 0]?5)
exten = s,2,SetGlobalVar(LOADBALANCE=0)
exten = s,3,Dial(${TRUNK_DENVER}/${ARG1})
exten = s,4,Goto(s,6)
exten = s,5,SetGlobalVar(LOADBALANCE=1)
exten = s,6,Dial(${TRUNK_DENVER2}/${ARG1})
exten = s,7,Goto(s,3)
--
Thank you and have a wonderful day,
Anthony Francis
Rockynet VOIP
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Re: [asterisk-users] Real API for Perl?
Alex Balashov wrote: Well, no, there really aren't any prebuilt high-level frameworks for approaching Asterisk through the Manager API or AGI. There is actually a couple of CPAN packages for interacting with the AMI in an event oriented fashion. http://search.cpan.org/search?query=asteriskmode=all Enjoy! -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP ___ -- 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] Problem with IRQ Share
Ruben Zamora wrote: Hi I have a Server with Centos 5, TDM400p, HP Server ML110. My problem is that I see IRQ Share with my TDM400P. How can I fix that??? ___ -- 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 man setpci ___ -- 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] pulling my hair out over voicemail
John Von Essen wrote: Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes to voicemail, and message is stored on server. I created an extension to retrieve the messages: exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain And that worked. Granted, everything is still defaults, so when I dial 1000, I get the Comedian Mail greeting, then it prompts for mailbox and password, then I get the menu. Now, here is how it gets weird. Today I go and setup a new second SIP phone, and proceed to set it up for voicemail. Inbound calls go to voicemail properly when nobody answers, but I cant retrieve the messages. When I dial extension 1000, its rings for 2 seconds, then just goes silent. No greeting, no mailbox prompts, nothing. Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 Where 6000 is my mailbox. But still nothing, when I dial 1000, it just goes silent. Please help. This is driving me nuts. I even tried re-installing asterisk from scratch - no change. -john ___ -- 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 I would suggest showing us the extensions configs for both phones :). ___ -- 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] chanspy does not pull the call back to asterisk after a reinvite
Franklin Webb wrote: Thanks to both of you for your input. I'll be in touch off list Steve. -Franklin - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 30, 2008 8:00:48 AM (GMT-0500) America/New_York Subject: Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite On Jan 29, 2008 8:36 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Jan 29, 2008 5:55 PM, Alex Balashov [EMAIL PROTECTED] wrote: Franklin, Because ChanSpy() is a passive monitor, there is nothing about the implementation that would cause Asterisk to shunt the speech back to itself. Asterisk only does this in situations where it is out of the media path and needs to insinuate itself back into it for the purpose of generating media, such as on-hold music, IVR, etc. What you're wanting should, in my opinion, basically be submitted as a feature request. Perhaps the developers can add a flag to the ChanSpy() invocation repertoire to make this work. Cheers, -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Alex, he was not asking why, it is obvious he knows why. He was asking for a solution or idea on how to work around this issue. Are you using Sangoma cards? If so, I might have a very good answer for you, as well as another very possible different solution. Both would be outside of Asterisk so some kind of magic would have to happen to associate the call being spied on to the channel but that should not be that difficult if you even need it. Another solution is to track down the code referenced here http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a reinvite back to asterisk before starting the spy. Anyways, I am sure it can be done. The question is how much time is it worth to make it happen. Maybe we should meet for lunch this week. I can meet you in cow country or Philly if you want, your choice. I have to go to both this week anyways and would like to catch up with things since Astricon. Thanks, Steve Totaro I just confirmed that there is a solution that is perfect for this that has been developed with a web interface to select the call to monitor. A little added code and you can pretty easily look up who the agent handling the call is. Let's test it out on your call center. Again, it is not an Asterisk app and would have no impact on your operations if it does not work. Thanks, Steve Totaro ___ -- 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 in sip.conf do canreinvite=no, and suddenly the audio is always available to asterisk. Anthony ___ -- 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] Peak number of calls?
Tilghman Lesher wrote: On Wednesday 23 January 2008 12:23:24 Gordon Henderson wrote: Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... No, the CDRs would be where that information is stored, if anywhere. This is actually sort of easy. You simply have every call pass through a context in which you assign the call to a group, then either do a NoOp echoing the group count or a user event doing the same, then either programatically or grep search your logs for the output or have a script monitoring the AMI watch for the user event and write the number in a data base. Of these two I personally do the second option because then I can just do a max() function on that database field to get the maximum calls for any time range I specify. Oh and just a note, never just say no because you don't know, in this instance you would say, I think your best bet is the CDR's. Just a tip. Anthony ___ -- 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] Your favorite Asterisk application.
Paul Hales wrote: I love writing dialplan, using vi. Does that make me weird? PaulH On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken ___ -- 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 I simply love vi, to the point which if an IDE doesn't have vi key bindings I loath using it. Anthony ___ -- 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] Anyone Using a Dell PowerEdge T105 in Production
Yeah I have several Dell SC 430-450's running Asterisk with a variety of Digium cards. I haven't been having a lot of those issues since CentOS 4.5. Scott Plante wrote: That link isn't working for me. Can you post some details and the price so I can see if I can recreate it? Thanks, Scott P.S. I got one of the new Vostro boxes a couple of months ago for a very small Asterisk install and CentOS wouldn't recognize the onboard network card. Given enough time we probably could have gotten it to work, but we ended up putting in a $10 network card and had no other trouble. The Vostro came in at $300 or so. P.S.S. I know a while back the Dell SC models had problems with Digium cards over interrupt handling. Anyone using SCs with Digium cards these days? - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 16 January 2008 19:39:34 o'clock (GMT-0500) America/New_York Subject: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production Unbeatable price for a low end Asterisk server (or any server for that matter) http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=bednv4ks=bsd http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=bednv4ks=bsd I wonder if anyone has any experience with this box and Digium or Sangoma hardware? Any compatibility issues? If not, I might stock up on them. Thanks, Steve Totaro ___ -- 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] Voicemail consultation problem
I would suppose that the time on the asterisk system is not the time that he is using. Other than that, you should really be collecting logs. David Florella wrote: Hello, A user who uses my Asterisk made me part of a worry about listening to his voicemails. He has received 4 voicemails on January 3, respectively at 3H00 pm, 3H36 pm, 3H41 pm and 4H40 pm. He has received notifications by e-mail at these times. On first listen to his messages, at 8.00 pm, Asterisk has announced two new voicemails(15H00 and 15H36). He has erased thos voicemails. At 8.30pm , he has called again the Asterisk voicemail. Asterisk announced him two messages (15H41 and 16H40). I don't have any Asterisk logs . A person have an idea of what may have caused the fact that my user did not, in the first call, heard his 4 messages? Thank you. ___ -- 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 -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [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] Developing Help
A good place to start is where any developer would start, at the site for the project they want to work on. http://asterisk.org/developers You can also get there by clicking on the giant link that says code for asterisk on the front page. Anthony Bhrugu Mehta wrote: hi, all, can anybody tell me how to be a part of asterisk developer team. I am so much intersted. thnks in advance. Bhrugu Mehta ___ -- 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] With rtcachefriends=yes, when do realtime changes take effect?
Adam Moffett wrote: I asked this question last week and never got an answer. I also didn't find the answer in the wiki. I think it would be nice if asterisk would check the database again if the user re-registers, but it doesn't seem to do that. A periodic update would be ok too, but it doesn't seem to do that either. It seems like changes never happen until a reload.if that is the case then doesn't rtcachefriends completely defeat the purpose of realtime SIP 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 New entries take effect immediately, however changes require a sip reload. ___ --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
