Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
On Fri, Oct 10, 2008 at 12:19:09AM -0500, Anthony Messina wrote: still, there are some concerning things that have been lingering, namely for me: http://bugs.digium.com/view.php?id=13443 This is the result of an incorrect sample file in asterisk 1.6.0, that wrongly uses a feature from 1.6.1 . The fixed sample file is the sole change of asterisk 1.6.0.1 . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] How to barge Inbound calls
Hi All Can anybody help me for dial plan which can barge inbound call groupwise. Because when i am trying to barge inbound calls which is coming on my DID number i can hear 1st 3 digit of my Inbound provider IP address instaed of extension which pick that calls. I tried Chanspy as well as Extenspy. But result is same. So Plz Help me. Thanks Amit ___ -- 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] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'
Sorry to post a C compile error on this mailing list but this is Asterisk related. Basically, I was following http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu e_logging to patch logger.c and Makefile in Asterisk 1.4.* in order to write queue_log to mySQL database. When I ran make, it complained: In function `write_mysql_logger': [...] /usr/src/asterisk-1.4.21.2/main/logger-mysql.c:98: undefined reference to `mysql_error' [...] collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 In my modified Makefile, I already had the line: ASTCFLAGS+=-I/usr/include/mysql and I found that mysql.h is already in /usr/include/mysql. I also already had mysql-client installed. In logger-mysql.c, there is already a line at the front of the program: #include mysql.h Any thoughts? ___ -- 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] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
On Thu, 9 Oct 2008, Steve Totaro wrote: I don't have answers just a question. DAHDI is alpha or beta code, what motivates you to upgrade so badly that you are frustrating yourself so much? Perhaps the fact that zaptel is not listed anymore on the Digium website? :) ___ -- 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] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
On Thu, 9 Oct 2008, Sean Bright wrote: On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse [EMAIL PROTECTED] wrote: The information (or lack of it) on upgrading from zaptel to that @*^QW%^%!!! dahdi is very frustrating. I cannot find anything on how to uninstall zaptel, i found an earlier post to this list which suggested make uninstall and make remove in the zaptel directory which just generates errors and does nothing (on zaptel 12.1). What types of errors do you encounter running 'make uninstall'? You'll need to make sure both asterisk and zaptel are shutdown before running make install: # service asterisk stop # service zaptel stop [EMAIL PROTECTED] dahdi-tools-2.0.0]# cd /usr/src/zaptel-1.4.12.1/ [EMAIL PROTECTED] zaptel-1.4.12.1]# make uninstall make: *** No rule to make target `uninstall'. Stop. [EMAIL PROTECTED] zaptel-1.4.12.1]# make remove make: *** No rule to make target `remove'. Stop. [EMAIL PROTECTED] zaptel-1.4.12.1]# Looking through the makefile there is only a target for make uninstall-modules which ofcourse only removes part of zaptel, not the init scripts and all the other stuff Unfortunately there was a bug in the initial 2.0.0 release. This has since been resolved in Subversion (see more details here http://bugs.digium.com/view.php?id=13615). If you'd like, you can grab the latest from Subversion of both the DAHDI Linux an DAHDI Tools packages, using the following commands: $ svn checkout http://svn.digium.com/svn/dahdi/linux/trunk dahdi-linux $ svn checkout http://svn.digium.com/svn/dahdi/tools/trunk dahdi-tools Also the config files and everything are much more complicated for dahdi than they were for zaptel As far as I am aware, the format of the configuration files (/etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf) are basically the same as their predecessors, /etc/zaptel.conf and /etc/asterisk/zapata.conf. Feel free to post here with any questions and we'll try to help out. OK, will do :) Thanks! ___ -- 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] Be aware of callcheap.com and Mike Low - It is scam
Hi everybody, Recently I was ripped off by this company named Callcheap Networks Inc, and so did one of the carriers I recommended to them. Now I am perusing legal action against them, a mess in which I never wanted to get into. Based on my bad experience, I wanted to let everybody know if this guy named Mike Low from Call Cheap Networks Inc. (callcheap.com) contacts you, please be careful. If you decide to do business with him, please get your money in advance and don't believe him on saying that he'd pay once the work is done or next week or tomorrow. He came to me from another company, with whom he hosted a callback service. He blamed them of bad service and asked me to setup a call back service for him. He seemed to be a genuine and honest person with very good plans for his business. He asked me for a good provider and I recommended him one. But when it came to pay for the service, he started to ask for more time. He did the same with the carrier. And after about month and a half, not paying to any one of us and using, or should I say misuing our trunks and services for his users and users of another company, he refused to pay us at all and blamed us for bad service, damage to his business and other non-sense. Once the carrier disconnect him after suffering a huge loss, and me refusing to do any further work for him unless my invoices are cleared, he threatening me to bring to court for disconnecting him, and is now looking for another company to host his callback service. During this month and a half, I also noticed that callcheap.com's website has changed its face and host three times. I have screen shots for last two for my record. If anybody of you have worked with this guy before, maybe they could share their experience here. And those who haven't, and if this guy contacts you, please be careful. -- Zeeshan A Zakaria ___ -- 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 enable inbound CLI for X-Lite/Asterisk phone.
Quoting Syed Nasruddin [EMAIL PROTECTED]:
I am using asterisk 1.4.18. I am using it for inbound only call center.
The SIP phones are X-Lite. Right now when a call is proxied by Asterisk
to X-Lite the agent only sees asterisk written on its CLI screen. I want
the agents to be able to view the callees number. Is there any way to
make this happen.
CLI showing as asterisk can indicate absent or withheld number. If
asterisk has it, it should pass it on to X-Lite without any special
settings.
Check to see if asterisk has CLI for the call by putting it in a NoOp in
the dialplan - NoOp(${CALLERID(all)}) would do. Watch asterisk with
verbose set to at least 3.
--
Phil Reynolds
mail: [EMAIL PROTECTED]
Web: http://www.tinsleyviaduct.com/phil/
Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
This message was sent using IMP, the Internet Messaging Program.
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Re: [asterisk-users] Compile logger-mysql.c with UNDEFINED REF to`mysql_error'
This looks really old and weird. I could suggest using realtime queue_log backport from 1.6 which i'm currently using. That's good info, Atis. I will definitely give it a go. winmail.dat___ -- 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] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
The reason for this is that 1.6.0 does not support dahdi. It was a mistake when it was listed as an included feature. The documentation for it has been removed in 1.6.0.1. If you need dahdi you need to go to 1.6.1. This is documented in this changelog: http://downloads.digium.com/pub/asterisk/ChangeLog-1.6.0.1 Brendan Martens On Oct 10, 2008, at 8:33 AM, Jim Duda wrote: Does anyone know what this error message means? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I've upgraded to 1.6.0 with dahdi 2.0. For some reason my outbound dahdi calls are not going through. At some point, it starts to work, but I don't know what the trigger is. Out of the blue, outbound calls start to work. I had been using asterisk-1.6-beta9 with zaptel without any problems. Thanks, Jim -- Executing [EMAIL PROTECTED]:1] Macro(SIP/111-b4e05610, dialout-dahdi,18005551212) in new stack -- Executing [EMAIL PROTECTED]:3] Set(SIP/111-b4e05610, DYNAMIC_FEATURES=outflash) in new stack -- Executing [EMAIL PROTECTED]:4] Dial(SIP/111-b4e05610, DAHDI/4/18005551212,40,tr) in new stack [Oct 10 08:29:10] WARNING[4365]: app_dial.c:1450 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/111-b4e05610, Dial Status:CHANUNAVAIL) in new stack -- Executing [EMAIL PROTECTED]:6] Goto(SIP/111-b4e05610, s-CHANUNAVAIL,1) in new stack -- Goto (macro-dialout-dahdi,s-CHANUNAVAIL,1) ___ -- 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] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
On Fri, Oct 10, 2008 at 3:22 AM, Remco Barendse [EMAIL PROTECTED] wrote: On Thu, 9 Oct 2008, Steve Totaro wrote: I don't have answers just a question. DAHDI is alpha or beta code, what motivates you to upgrade so badly that you are frustrating yourself so much? Perhaps the fact that zaptel is not listed anymore on the Digium website? :) Well it is listed like its production code. Not sure it is. I have a Digium Wildcard TE110P. I recently upgraded from 1.4.21.2 and zaptel to 1.4.22 and dahdi. The upgrade seemed to go well, downed asterisk, used the /etc/init.d script to down zaptel, made change to asterisk.conf about still using ZAP names (will take a while even with sed as it is any combination of zap/ZAP/Zap throughout many exten.*.conf files), removed the old kernel module, ran dahdi script (which installed new kernel module), brought up asterisk. All worked (note that I didn't reboot the system - it has been up for about 60 days continuous). Two days ago work on the power grid in the building demanded a shutdown of the system. Coming up, asterisk ran, but never could get dahdi working. Everything from kernel panics, to system lockups, etc. The main module loaded, but when the echocanceller loaded ... After about 2 hours fighting and resetting the server any number of times, I ripped out dahdi and asterisk-1.4.22, reinstalled the old zaptel and asterisk-1.4.21.2 and all is well. Dahdi is definitely _not_ ready for prime time. I've seen kernel panics on other Digium hardware as well and was basically told it was my hardware. Funny I don't get those kernel panics using zaptel. And yes, you just have to figure out that you need to copy genconf_parameters from the dahdi/xpp directory to /etc/dahdi -- it's not automagic. Ciao, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto ___ -- 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] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
Brendan Martens wrote: The reason for this is that 1.6.0 does not support dahdi. It was a mistake when it was listed as an included feature. The documentation for it has been removed in 1.6.0.1. If you need dahdi you need to go to 1.6.1. That is incorrect. There was one small feature (the 'dahdichan' configuration option, used when creating a new-style chan_dahdi.conf instead of the format used by zapata.conf) that was mistakenly included in the sample config file in 1.6.0, which has how been removed in 1.6.0.1. The dahdichan config option is supported in 1.6.1. 1.6.0 fully supports DAHDI, and does not support Zaptel at all. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
I see, Thank you for the clarification. Brendan Martens On Oct 10, 2008, at 9:31 AM, Kevin P. Fleming wrote: Brendan Martens wrote: The reason for this is that 1.6.0 does not support dahdi. It was a mistake when it was listed as an included feature. The documentation for it has been removed in 1.6.0.1. If you need dahdi you need to go to 1.6.1. That is incorrect. There was one small feature (the 'dahdichan' configuration option, used when creating a new-style chan_dahdi.conf instead of the format used by zapata.conf) that was mistakenly included in the sample config file in 1.6.0, which has how been removed in 1.6.0.1. The dahdichan config option is supported in 1.6.1. 1.6.0 fully supports DAHDI, and does not support Zaptel at all. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
Inbound calls on DAHDI work fine. At some point, outbound starts working. I just cannot figure out what the trigger is. At first, I thought the trigger was receiving at least one inbound call. But that isn't always true. Once it starts working, it seems to continue until a restart. Everything looks normal. I'm having trouble with outbound calls on channel 4 (attached to the PSTN). asterisk*CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudointernal default In Service 1internal default In Service 4incoming default In Service asterisk*CLI dahdi show channel 4 Channel: 4LI File Descriptor: 21 Span: 1 Extension: Dialing: no Context: incoming Caller ID: Calling TON: 0 Caller ID name: Mailbox: 100 Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook Jim Kevin P. Fleming wrote: Brendan Martens wrote: The reason for this is that 1.6.0 does not support dahdi. It was a mistake when it was listed as an included feature. The documentation for it has been removed in 1.6.0.1. If you need dahdi you need to go to 1.6.1. That is incorrect. There was one small feature (the 'dahdichan' configuration option, used when creating a new-style chan_dahdi.conf instead of the format used by zapata.conf) that was mistakenly included in the sample config file in 1.6.0, which has how been removed in 1.6.0.1. The dahdichan config option is supported in 1.6.1. 1.6.0 fully supports DAHDI, and does not support Zaptel at all. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodríguez ___ -- 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] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
Anthony Messina wrote: On Thursday 09 October 2008 09:57:30 pm Steve Totaro wrote: Now I have not touched any of that code, but to me, it would have been much simpler to change names, then change functionality later. Make DAHDI a drop in replacement for Zaptel, in fact, if memory serves me correctly that is what someone at Digium explained, it was merely a find and replace operation. i agree with the idea that a drop in should have been created, and functionality built from there. Hindsight being 20/20, this may have been better. Although on the off chance that anyone is interested, the line of thought was to lump in all the changes that would require people to touch their configuration files, be it name changes, layouts, etc.., in order to reduce the number of times they had to think about their Zaptel / DAHDI configuration files. If you're already going through the process of renaming your files and scripts, might as well make all the changes when you upgrade as opposed to keep having to make modifications in stages. That was the thought anyway. ___ -- 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 7960 sccp, Skinny and 1.4
You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Thank you, Wayne. ___ -- 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] Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
Does anyone know what this error message means? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I've upgraded to 1.6.0 with dahdi 2.0. For some reason my outbound dahdi calls are not going through. At some point, it starts to work, but I don't know what the trigger is. Out of the blue, outbound calls start to work. I had been using asterisk-1.6-beta9 with zaptel without any problems. Thanks, Jim -- Executing [EMAIL PROTECTED]:1] Macro(SIP/111-b4e05610, dialout-dahdi,18005551212) in new stack -- Executing [EMAIL PROTECTED]:3] Set(SIP/111-b4e05610, DYNAMIC_FEATURES=outflash) in new stack -- Executing [EMAIL PROTECTED]:4] Dial(SIP/111-b4e05610, DAHDI/4/18005551212,40,tr) in new stack [Oct 10 08:29:10] WARNING[4365]: app_dial.c:1450 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/111-b4e05610, Dial Status:CHANUNAVAIL) in new stack -- Executing [EMAIL PROTECTED]:6] Goto(SIP/111-b4e05610, s-CHANUNAVAIL,1) in new stack -- Goto (macro-dialout-dahdi,s-CHANUNAVAIL,1) ___ -- 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] How to enable inbound CLI for X-Lite/Asterisk phone.
Hi, I am using asterisk 1.4.18. I am using it for inbound only call center. The SIP phones are X-Lite. Right now when a call is proxied by Asterisk to X-Lite the agent only sees asterisk written on its CLI screen. I want the agents to be able to view the callees number. Is there any way to make this happen. Regards Syed Nasruddin ___ -- 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] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'
On Fri, Oct 10, 2008 at 10:50 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Sorry to post a C compile error on this mailing list but this is Asterisk related. Basically, I was following http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu e_logging to patch logger.c and Makefile in Asterisk 1.4.* in order to write queue_log to mySQL database. When I ran make, it complained: In function `write_mysql_logger': [...] /usr/src/asterisk-1.4.21.2/main/logger-mysql.c:98: undefined reference to `mysql_error' [...] collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 In my modified Makefile, I already had the line: ASTCFLAGS+=-I/usr/include/mysql and I found that mysql.h is already in /usr/include/mysql. I also already had mysql-client installed. In logger-mysql.c, there is already a line at the front of the program: #include mysql.h Any thoughts? This looks really old and weird. I could suggest using realtime queue_log backport from 1.6 which i'm currently using. http://ftp.iq-labs.net/queue_log-1.4/asterisk_queue_log_realtime_1.4.19.patch This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] Hang up detection with TDM400P and Telewest/Virgin Media line
On Thu, 9 Oct 2008, Mike wrote: On Fri, Oct 10, 2008 at 08:10:39AM +1930, Luis Morales wrote: Mike, Can you tell us : - asterisk version - zaptel version When you call over this line, when you hangup did you hear an busy tone ? or any class tone ? To do this test connect your lines to analog phone and make a call. Let's us know the results. Regards, Luis Morales Zaptel Version: 1.2.11 Asterisk 1.2.13 I called my mobile from the line and hung up. The line just went silent. There were no tones. I also watched the lamp on the phone, it didn't got out. I guess this could be because the line current isn't dropped or maybe because of capacitance in the phone? Best way is to watch via the CLI - set verbose 3 and watch what happens. I tried this on my BT line and when I clear down, the lamp on the phone goes off momentarilly and then I get a single, continuous tone. Gordon, would you mind doing this test on your line to see what happens? I'll do it tonight (or probably tomorow) - both sites are live running today and I'll need to do it via a remote IAX phone as I'm not driving to each site to do the test. Gordon ___ -- 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] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
Steve Totaro wrote: On Thu, Oct 9, 2008 at 10:32 PM, sean darcy [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Remco Barendse wrote: The information (or lack of it) on upgrading from zaptel to that @*^QW%^%!!! dahdi is very frustrating. I cannot find anything on how to uninstall zaptel, i found an earlier post to this list which suggested make uninstall and make remove in the zaptel directory which just generates errors and does nothing (on zaptel 12.1). Then i install dahdi-linux and dahdi-tools and i want to start configuring it, so i am trying dahdi_genconf like the docs suggested which generates this really helpful error message : /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No such file or directory Also the config files and everything are much more complicated for dahdi than they were for zaptel There was some nice documentation and examples on how to get started with configuring certain devices with zaptel on the digium page, for my TDM11B they only mention zaptel. Did anyone even try this? It'll work. But it's not easy. I didn't find dahdi_genconf helpful. Post your /etc/dahdi/system.conf ( the analogue of zaptel.conf ) and /etc/asterisk/chan_dahdi.conf ( analogue of zapata.conf ). With some help, you'll fix this. sean Total hindsight and thinking as a user, but the initial explanation of DAHDI came out because someone put something out there premature and someone noticed that Zaptel was being replaced by DAHDI. The party line explanation from Digium was that someone owned the rights to the zaptel name. A calling card dealer who had been very nice to allow Digium to continue using the Zaptel name but was at his end, so hence the name change. This *is* the correct reason. Not sure I totally buy that but whatever, my thought was it was to remove any rights or credits from the Zapata Telephony Project and Jim Dixon. Digium could control DAHDI exactly the way it controls Asterisk, Jim's name is still on the source code, and still intentionally is there. Please don't jump to any rash conclusions. You can certainly still use Zaptel as Zaptel if you'd like. We were forced to change it due to the name related issues that have been mentioned. We're just grateful that the other party that brought the issue up has been so patient since it has taken so long. It has been a bit of a rocky road with some of the new features that were put it into it, but, any time you rewrite code or do something new, there's always going to be a period of shaking out of unforeseen bugs. Sorry if you have had any trouble. The name change and related efforts have been just as hard on us as developers as it has been on people that use it. -- Matthew Fredrickson Software/Hardware Engineer Digium, Inc. ___ -- 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] Howto analyze concurrent ISDN channel usage
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Thursday, October 09, 2008 2:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Howto analyze concurrent ISDN channel usage Hi, Does anyone have a suggestion how I can analyze the concurrent usage of ISDN channels? A client complains about their clients sometimes getting a busy tone when trying to call them. I suspect they just need to add an additional ISDN2 line but I need some conclusive information that they are indeed maxing out their ISDN channels. Thanks, Patrick ___ -- 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 -- Hi, It may not be that you are out of channels. I've recently tried to setup my ISDN line for use with asterisk and ran into a similar issue. Some people could call me and others couldn't. My asterisk box was rejecting some calls with an Incompatible Destination Cause code 88. I found that some phone lines/numbers just couldn't call my isdn line. I still haven't figured it out yet... Jay ___ -- 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 SIP calls stop working having more than 300 calls (more than 600 channels)
Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Any suggestion?? On Fri, Oct 10, 2008 at 11:37 AM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodríguez Juan, You might need to increase the number of file descriptors available in Linux. What distro are you on? Are you using the Asterisk startup scripts? In later versions this is done for you automatically if you are running Asterisk as root. Have a look at this: http://www.voip-info.org/wiki/view/file+descriptors -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- 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 SIP calls stop working having more than 300 calls (more than 600 channels)
On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: After getting some ERRORS like this: [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call. [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call. I start getting: ERROR[14844] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error) [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for 'TRUNK/DESTINATION' (Out of memory or socket error). I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured. Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call. The configuration of the * is an incomming call from the my SIP Provider and after internal manage it makes a second call to other destination--DID--. For AGI compatibility issues I could not use Version 1.4.22 (issues whith DeadAGI for billing purpuses). This is my rtp.conf [general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 ; rtpstart=1 rtpend=2 This is my sip.conf for the TRUNK [TRUNK] type=peer nat=never host=destination.public.ip.address fromdomain=my.public.ip.address dtmfmode=rfc2833 canreinvite=no disallow=all allow=g729 Please help. -- Juan E. Rodríguez Juan, You might need to increase the number of file descriptors available in Linux. What distro are you on? Are you using the Asterisk startup scripts? In later versions this is done for you automatically if you are running Asterisk as root. Have a look at this: http://www.voip-info.org/wiki/view/file+descriptors -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about echo cancelation
Hi, I'm using the following setup : Alice IPPhone --LAN- Media gateway PSTN --- Phone Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably, when users complains about earing their own voice, that means that distant phone or nearby equipment is leaking : Bob's phone is sending Alice's voice signal back to Alice. So, to properly cancel, I would say Media gateway should substract from incoming signal the signal that left the media gateway few ms before. Discussing here and there, some say that Media Gateway never work this way : it would only filters out locally generated echo. Do you agree with that ? If positive, then what can you do, if Bob's phone generate much echo ? Regards ___ -- 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] softclient for customized apps like a call center?
There are a variety of open source (and closed source) software-based Windows SIP or IAX phones out there. However, I am thinking of using one for a inbound call center. Some of the things I'd be looking for: The ability to make/receive calls (duh!). The ability to 'launch' a web browser based on incoming call conditions. For example, launching an URL like: http://mydomain.com/lookupcustomer.php?cnam=xx BTW, I'd be happy to write the ability in myself, in which case the question is: is the source code easily modifiable? The ability to create special buttons for unique functions. The ability to make MySQL queries or HTTP queries for some kind of status screen? Sort of like what many of hard phones allow. Redundant registrations. While I am more than willing to consider closed-source stuff, I somehow doubt those solutions are customizable enough. Suggestions/Opinions? ___ -- 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 SIP calls stop working having more than 300 calls (more than 600 channels)
On 10/10/08, Juan Rodríguez [EMAIL PROTECTED] wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Any suggestion?? Here's another (fuller) list, shamelessly lifted from another mailing list: ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 99 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited Make sure these are in your Asterisk startup scripts before Asterisk starts. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Got event 17 (Polarity Reversal)...
Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]: chan_dahdi.c:7379 mwi_thread: Got event 17 (Polarity Reversal)... Passing along to ss_thread -- Starting simple switch on 'DAHDI/4-1' [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 4 (Alarm)... [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 17 (Polarity Reversal)... [Oct 10 12:47:56] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 5 (No more alarm)... -- Executing [EMAIL PROTECTED]:1] Goto(DAHDI/4-1, incoming-dial,s,1) in new stack ___ -- 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] Block Caller ID
Hi Is there any way to stop Asterisk from sending Caller ID display on the softphones ? I;ve E1 PRIs and SIP extensions , i need to stop caller ID from appearing on the softphones ...but in CDRs caller Ids should show - so please dont suggest to set blockcallerid=yes in zapata.conf ;) Thanks Sriram ___ -- 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] Transfer/Park Question.
Daniel Hazelbaker wrote: On Oct 9, 2008, at 2:59 PM, Brent Davidson wrote: Short answer: currently no. Medium answer: I just rolled out 60+ Snom phones (300s and 320s) and we do call parking with DTMF. People were used to just hitting PARK and their phone displaying the park extension (old NEC system). I didn't tell anybody anything except it will speak the extension back to you and nobody has complained about hearing the DTMF digits. We chose a 3 digit code (#92 I believe) to try an alleviate the possibility of somebody accidently parking a call while filling out a DTMF based form/menu system, but in theory you could assign just * to park and only deal with 1 tone. Just be aware that if the user needs to hit * for anything else, they won't be able to use it. Long answer: Snom phones support text messages to the phone that automatically display. I am looking for a way to use that in conjunction with Snom's ParkOrbit feature (which does work, you just don't hear the extension). Basically Asterisk would do a normal park and then trigger a SIP NOTIFY message to the parkING phone that says Parked: 701. The message can be cleared by the user by pressing X, or ideally Asterisk would auto-clear the message after 10 seconds (or whatever). In theory I can do the long answer now with a Manager application, but I don't like the idea of relying on an external application. If it crashes or locks up for whatever reason then suddenly people get parked and nobody knows where. Also be aware that in 1.2.x and 1.4.x, if you park a call and then pick it up, you can't park it again. At least not with the DTMF method. I borrowed a patch from the 1.6 branch that fixes this and made it applicable to 1.4.20.1, well I borrowed part of it. The entire patch let you configure who could park etc., I wanted both sides to always park so I just took the 2 or 3 lines that were needed for that. If you are interested I can e-mail it to you directly. Regards, Daniel I wasn't aware of the inability to re-park calls in 1.4 That could have been a nasty surprise. I would be very interested in the patch that fixes that. Thanks, Brent ___ -- 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] Got event 17 (Polarity Reversal)...
You should not get that message on analog lines in the USA or Canada. I suspect your line has a provisioning issue or is using different signaling than you think it is using. Jim Duda wrote: Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]: chan_dahdi.c:7379 mwi_thread: Got event 17 (Polarity Reversal)... Passing along to ss_thread -- Starting simple switch on 'DAHDI/4-1' [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 4 (Alarm)... [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 17 (Polarity Reversal)... [Oct 10 12:47:56] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 5 (No more alarm)... -- Executing [EMAIL PROTECTED]:1] Goto(DAHDI/4-1, incoming-dial,s,1) in new stack ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- 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] Be aware of callcheap.com and Mike Low - It is scam
On Fri, Oct 10, 2008 at 5:09 AM, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi everybody, Recently I was ripped off by this company named Callcheap Networks Inc, and so did one of the carriers I recommended to them. Now I am perusing legal action against them, a mess in which I never wanted to get into. Based on my bad experience, I wanted to let everybody know if this guy named Mike Low from Call Cheap Networks Inc. (callcheap.com) contacts you, please be careful. If you decide to do business with him, please get your money in advance and don't believe him on saying that he'd pay once the work is done or next week or tomorrow. He came to me from another company, with whom he hosted a callback service. He blamed them of bad service and asked me to setup a call back service for him. He seemed to be a genuine and honest person with very good plans for his business. He asked me for a good provider and I recommended him one. But when it came to pay for the service, he started to ask for more time. He did the same with the carrier. And after about month and a half, not paying to any one of us and using, or should I say misuing our trunks and services for his users and users of another company, he refused to pay us at all and blamed us for bad service, damage to his business and other non-sense. Once the carrier disconnect him after suffering a huge loss, and me refusing to do any further work for him unless my invoices are cleared, he threatening me to bring to court for disconnecting him, and is now looking for another company to host his callback service. During this month and a half, I also noticed that callcheap.com's website has changed its face and host three times. I have screen shots for last two for my record. If anybody of you have worked with this guy before, maybe they could share their experience here. And those who haven't, and if this guy contacts you, please be careful. -- Zeeshan A Zakaria http://lists.digium.com/mailman/listinfo/asterisk-users While pending legal action, it is best to not post these things in a public forum. While I appreciate the fact you are trying to help the community by giving a heads up, you set yourself up for a counter suit for defamation of character. It is best to wait for court to rule before going public. At the very least, use a hushmail account and post a warning with no identifiable transactions or actions, just watch out for so and so, although I still think posting one sided things to a list serves nobody well, if they have no way to address the accusations, and like I said before, it is best to do it court. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Question about echo cancelation
All calls with a 2-wire analog piece have echo. You cannot perceive the echo because it happens so fast on non-VoIP connections. On VoIP calls you have significant extra latency while causes you you to perceive the echo. Echo must be removed before the call is converted to VoIP -- in your case the Media Gateway is the device that must remove echo. Olivier wrote: Hi, I'm using the following setup : Alice IPPhone --LAN- Media gateway PSTN --- Phone Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably, when users complains about earing their own voice, that means that distant phone or nearby equipment is leaking : Bob's phone is sending Alice's voice signal back to Alice. So, to properly cancel, I would say Media gateway should substract from incoming signal the signal that left the media gateway few ms before. Discussing here and there, some say that Media Gateway never work this way : it would only filters out locally generated echo. Do you agree with that ? If positive, then what can you do, if Bob's phone generate much echo ? Regards ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- 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] Howto analyze concurrent ISDN channel usage
Luis Morales wrote: Try with fop, http://www.asternic.org/ Thanks Luis. I'll give that a try. Regards, Patrick ___ -- 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] Howto analyze concurrent ISDN channel usage
Jay Taylor wrote: Hi, It may not be that you are out of channels. I've recently tried to setup my ISDN line for use with asterisk and ran into a similar issue. Some people could call me and others couldn't. My asterisk box was rejecting some calls with an Incompatible Destination Cause code 88. I found that some phone lines/numbers just couldn't call my isdn line. I still haven't figured it out yet... Thanks for the info Jay. Do you use bristuff by any chance? Regards, Patrick ___ -- 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] Got event 17 (Polarity Reversal)...
On Fri, Oct 10, 2008 at 12:57:27PM -0400, Jim Duda wrote: Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. grep polarity /etc/asteirsk/chan_dahdi.conf See the documentation for answeronpolarityswitch and hanguponpolarityswitch . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 SIP calls stop working having more than 300 calls (more than 600 channels)
On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodríguez wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Are file descriptors the problem? ls /proc/PID_OF_ASTERISK/fd | wc Maybe there are really not enough open ports? Start with: netstat -anu Or: netstat -anup -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Howto analyze concurrent ISDN channel usage
Stefan Schmidt wrote: you could use mrtg to get stats of the overall usage of the server. or Thanks for your suggestion. I found a script here: http://karlsbakk.net/asterisk/ Regards, Patrick ___ -- 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] Transfer/Park Question.
Brent Davidson wrote:
Also be aware that in 1.2.x and 1.4.x, if you park a call and then
pick it up, you can't park it again. At least not with the DTMF
I wasn't aware of the inability to re-park calls in 1.4 That could have
been a nasty surprise. I would be very interested in the patch that
fixes that.
I don't remember where I got it (Might have been the bug tracker) that
works fine under the current 1.4.x. I had to do a minor change to get
it to apply.
Copy into Asterisk source directory
patch -p0 *.patch
rm *.patch
make
make install
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
Index: res/res_features.c
===
--- res/res_features.c (revision 84405)
+++ res/res_features.c (working copy)
@@ -1670,7 +1670,7 @@
}
if (con) {
char returnexten[AST_MAX_EXTENSION];
- snprintf(returnexten, sizeof(returnexten), %s|30|t, peername);
+ snprintf(returnexten, sizeof(returnexten), %s|30|tk, peername);
ast_add_extension2(con, 1, peername, 1, NULL, NULL, Dial, strdup(returnexten), ast_free, registrar);
}
set_c_e_p(chan, parking_con_dial, peername, 1);
@@ -1927,6 +1927,7 @@
memset(config, 0, sizeof(struct ast_bridge_config));
ast_set_flag((config.features_callee), AST_FEATURE_REDIRECT);
ast_set_flag((config.features_caller), AST_FEATURE_REDIRECT);
+ ast_set_flag((config.features_caller), AST_FEATURE_PARKCALL);
res = ast_bridge_call(chan, peer, config);
pbx_builtin_setvar_helper(chan, PARKEDCHANNEL, peer-name);
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Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
On 08:26, Fri 10 Oct 08, David Gibbons wrote: You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Or, if you dont want to use outside modules use Asterisk 1.6 (which has been released as well) with the chan_skinny driver. A lot of development went into it and it's much more useable then the 1.2 version. Myself uses chan_skinny in production without too much trouble. Specially when you use the 7960 phones it's a nice setup. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. What driver are you referring to ? It must be something outside of the core asterisk, because a lot of commits went into chan_skinny the last year or so. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Yes, chan_skinny in 1.4 is better then the 1.2 version, but the real stuff happened in the 1.6 version. 1.6.0 is released, so why not use that one instead of 1.4? Thank you, Wayne. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Got event 17 (Polarity Reversal)...
You should not get that message on analog lines in the USA or Canada. I suspect your line has a provisioning issue or is using different signaling than you think it is using. Not necessarily true. Most recent solid state switches have abandoned this as a cost saving measure. Polarity reversal was originally done on all loop start and ground start circuits on the calling end when the called party answered. It was used to identify start of call. On a relay based step switch it was fairly easy to do, because the path went through a relay contact that changed at that point anyway, so it was just deliberately wired backwards. It created havoc with early DTMF pads, because they used transistors and needed a known polarity. That was solved by adding a bridge rectifier in the dialpad. Loop and Ground Start trunks can still be ordered with this option because it helps accurately determine start of call for CDR. However, it requires a more expensive line card in the CO, so they don't like to do it. A similar switch when a call is first terminated (sometimes before it rings) can be used to insure that a line is not accidentally picked for an outgoing call by a PBX when a call is coming in on it. Wilton ___ -- 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 SIP calls stop working having more than 300 calls (more than 600 channels)
Having 600 channels it would be like 1200 RTP ports. And on the rtp.conf I have fonfigured from 1 to 2. I do not think this is the problem. Thanks, Juan On Fri, Oct 10, 2008 at 1:38 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Fri, Oct 10, 2008 at 11:56:34AM -0400, Juan Rodríguez wrote: Kristian: Thanks for your reply. I am running asterisk as root, but still getting this error. I did a test while running asterisk 1.4.21 version setting ulimit -n 32768, but after restaring asterisk it stop working with less than 150 calls (less than 300 channels). Are file descriptors the problem? ls /proc/PID_OF_ASTERISK/fd | wc Maybe there are really not enough open ports? Start with: netstat -anu Or: netstat -anup -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 -- Juan E. Rodríguez ___ -- 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] Got event 17 (Polarity Reversal)...
Hi Jim, We had this exact problem with our system for several years. A call would come in with no caller ID and when we answered nobody would be there. On the Asterisk console would be the Got event 17 (Polarity Reversal) message. We spent hours and hours on this. Our carrier was ATT (SBC). I spoke with their tech support several times and spoke with many phone guys. Nobody could solve it. We moved our office to a new location and are now using Comcast VOIP lines into the TDM card and the problem is gone now. It had to be something with the POTS lines we were getting from ATT. Another interesting thing is that we didn't get the calls all the time but when we did get them they were ALWAYS on the hour or half-hour. Thanks, Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Duda Sent: Friday, October 10, 2008 11:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Got event 17 (Polarity Reversal)... Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]: chan_dahdi.c:7379 mwi_thread: Got event 17 (Polarity Reversal)... Passing along to ss_thread -- Starting simple switch on 'DAHDI/4-1' [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 4 (Alarm)... [Oct 10 12:47:55] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 17 (Polarity Reversal)... [Oct 10 12:47:56] NOTICE[6671]: chan_dahdi.c:7114 ss_thread: Got event 5 (No more alarm)... -- Executing [EMAIL PROTECTED]:1] Goto(DAHDI/4-1, incoming-dial,s,1) in new stack ___ -- 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] Got event 17 (Polarity Reversal)...
Dan Peters wrote: interesting thing is that we didn't get the calls all the time but when we did get them they were ALWAYS on the hour or half-hour. Sounds like it may have been a line test. I vaguely recall a thread going on here about such tests causing issues. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,n,Voicemail([EMAIL PROTECTED])
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,101,1)
[outgoing]
; The following gives an Unknown Caller ID
;exten = _1NXXNXX,1,Set(CALLERID(num)=XX)
;exten = _1NXXNXX,2,Set(CALLERID(name)=XX)
exten = _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _1NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXX,1,Set(CALLERID(num)=9045622082)
exten = _NXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
Okay I'm going to start simple.
First I would like to forward the number to the remote number which
we'll make 904-940-9007. I've commented out the voicemail for the time
being, I'll bring that in once a menu is composed later on. So anyways
I've added a second rule to dial the second number after 20 seconds is
that the correct placement?
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten = 101,1,Dial(SIP/101,20)
exten = 101,n,Dial(SIP/[EMAIL PROTECTED])
;exten = 101,n,Voicemail([EMAIL PROTECTED])
;This automatically calls the right mailbox using the ${CALLERIDNUM}
variable in the current context (var ${CONTEXT}).
exten=*98,1,VoiceMailMain([EMAIL PROTECTED])
include = inbound
include = outgoing
[inbound]
exten = 9045622082,1,Goto(default,101,1)
[outgoing]
; The following gives an Unknown Caller ID
;exten = _1NXXNXX,1,Set(CALLERID(num)=XX)
;exten = _1NXXNXX,2,Set(CALLERID(name)=XX)
exten = _1NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _1NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _1NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXX,1,Set(CALLERID(num)=9045622082)
exten = _NXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _NXXNXX,1,Set(CALLERID(num)=9045622082)
exten = _NXXNXX,n,Set(CALLERID(name)=Stephen Reese)
exten = _NXXNXX,n,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Set(CALLERID(num)=9045622082)
exten = _011.,n,Set(CALLERID(name)=Stephen Reese)
exten = _011.,n,Dial(SIP/[EMAIL PROTECTED])
exten = _911,1,Set(CALLERID(num)=9045622082)
exten = _911,n,Set(CALLERID(name)=Stephen Reese)
exten = _911,n,Dial(SIP/[EMAIL PROTECTED])
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Re: [asterisk-users] Transfer/Park Question.
Doug Lytle wrote: Brent Davidson wrote: Also be aware that in 1.2.x and 1.4.x, if you park a call and then pick it up, you can't park it again. At least not with the DTMF I wasn't aware of the inability to re-park calls in 1.4 That could have been a nasty surprise. I would be very interested in the patch that fixes that. I don't remember where I got it (Might have been the bug tracker) that works fine under the current 1.4.x. I had to do a minor change to get it to apply. Copy into Asterisk source directory patch -p0 *.patch rm *.patch make make install Doug Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf and none of them seem to work when manually dialed or when bound as a DTMF code to a key. So far I've tried the following under [featuremap] in features.conf: parkcall = *5 parkcall = #72 parkcall = *9 parkcall = #75 I don't even see any acknowledgment of the DTMF tones showing up on the console. ___ -- 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] Transfer/Park Question.
Brent Davidson wrote: Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf Check your featuredigittimeout, it defaults to 1/2 second. You may need to increase it. I have mine set to ## to activate, easier to do it quickly. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Budge Tones pick up wrong calls
We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For example, 2501 might respond to the calls for 2518. After a reboot, it might decide to respond to 2501 as it should. Or it might respond to 2536. The phone it responds for will not respond. I don't know whether to look in the settings on the phone or in an Asterisk setting, and what setting to check in either place. Has anyone seen this behavior before? --Paul -- Paul Douglas Franklin Computer Manager, Union Gospel Mission of Yakima, Washington Husband of Danette Father of Laurene, Miriam, Tycko, Timothy, Sarabeth, Marie, Dawnita, Anna Leah, Alexander, and Caleb ___ -- 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] Transfer/Park Question.
On Oct 10, 2008, at 1:00 PM, Brent Davidson wrote: Doug Lytle wrote: I don't remember where I got it (Might have been the bug tracker) that works fine under the current 1.4.x. I had to do a minor change to get it to apply. Copy into Asterisk source directory patch -p0 *.patch rm *.patch make make install Doug Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf and none of them seem to work when manually dialed or when bound as a DTMF code to a key. So far I've tried the following under [featuremap] in features.conf: parkcall = *5 parkcall = #72 parkcall = *9 parkcall = #75 I don't even see any acknowledgment of the DTMF tones showing up on the console. You won't. The patch I sent you off-list is incomplete, this one is better. I forgot I fixed the parked has timed out option in another patch before I fixed this part. Anyway, make sure when you dial you put k in the dial options (K too if you want both sides to park). It used to be tied to the t option I believe and then got moved out to k at some point. Other than that, it should work. Daniel ___ -- 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] Cisco 7960 sccp, Skinny and 1.4
Thanks both, The only thing I have a little concern over is that 1.6 is that its still a development release (if I understand things correctly). Stability is the main thing for me (its only a very small set up) but there are no technical people around if something were to go wrong through the day. I shall take another look at both options. Thank you Wayne. Michiel van Baak wrote: On 08:26, Fri 10 Oct 08, David Gibbons wrote: You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Or, if you dont want to use outside modules use Asterisk 1.6 (which has been released as well) with the chan_skinny driver. A lot of development went into it and it's much more useable then the 1.2 version. Myself uses chan_skinny in production without too much trouble. Specially when you use the 7960 phones it's a nice setup. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. What driver are you referring to ? It must be something outside of the core asterisk, because a lot of commits went into chan_skinny the last year or so. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Yes, chan_skinny in 1.4 is better then the 1.2 version, but the real stuff happened in the 1.6 version. 1.6.0 is released, so why not use that one instead of 1.4? Thank you, Wayne. ___ -- 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] Transfer/Park Question.
Doug Lytle wrote: Brent Davidson wrote: Ok, the patch is working great. Any idea what would make the one step parking not work? I've tried several DTMF combinations in features.conf Check your featuredigittimeout, it defaults to 1/2 second. You may need to increase it. I have mine set to ## to activate, easier to do it quickly. Doug I checked that. I've got mine set to 800 and all of my other 2-digit features work (transfer, blind transfer, etc). The only one that doesn't is the parkcall feature. ___ -- 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] Transfer/Park Question.
Daniel Hazelbaker wrote: You won't. The patch I sent you off-list is incomplete, this one is better. I forgot I fixed the parked has timed out option in another patch before I fixed this part. Anyway, make sure when you dial you put k in the dial options (K too if you want both sides to park). It used to be tied to the t option I believe and then got moved out to k at some point. Other than that, it should work. Daniel That was it. Needed to add the k options. All is working now. Also, I don't think I got a patch from you off-list. The one I got from you was the asterisk-1.4.20.1-callreparking.patch. Got another one from Doug Lytle called multi-park.patch. ___ -- 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] Help need for debuging the core file.
I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. # thread apply all bt Thread 6 (process 20135): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb7469b80, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb7469b4c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xb22bf9a8, c1=0xa2ae648, config=0xb746a7a0, fo=0xb7469c40, rc=0xb7469c44) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xb22bf9a8, peer=0xa2ae648, config=0xb746a7a0) at res_features.c:1365 #5 0x005a40ba in dial_exec_full (chan=0xb22bf9a8, data=Variable data is not available. ) at app_dial.c:1633 #6 0x005a6a33 in dial_exec (chan=0xfffc, data=0x7fff) at app_dial.c:1680 #7 0x08090bad in pbx_extension_helper (c=0xb22bf9a8, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xb22bf9a8) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 5 (process 11504): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb51e2e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb51e2e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xa2db720, c1=0xa12acf8, config=0xb51e37c0, fo=0xb51e2f50, rc=0xb51e2f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xa2db720, peer=0xa12acf8, config=0xb51e37c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb51e3ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0xa2db720, data=0xb51e8010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0xa2db720, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0xa2db720) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 ---Type return to continue, or q return to quit--- #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 4 (process 24033): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002dfdf4 in poll () from /lib/tls/libc.so.6 #2 0x080675a7 in ast_waitfor_nandfds (c=0xb6c56e90, n=2, fds=0x0, nfds=0, exception=0x0, outfd=0x0, ms=0xb6c56e5c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0x9e9cb80, c1=0xaf0ce2f8, config=0xb6c577c0, fo=0xb6c56f50, rc=0xb6c56f54) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0x9e9cb80, peer=0xaf0ce2f8, config=0xb6c577c0) at res_features.c:1365 #5 0x004085bb in try_calling (qe=0xb6c57ac0, options=Variable options is not available. ) at app_queue.c:2602 #6 0x0040c6eb in queue_exec (chan=0x9e9cb80, data=0xb6c5c010) at app_queue.c:3344 #7 0x08090bad in pbx_extension_helper (c=0x9e9cb80, con=Variable con is not available. ) at pbx.c:574 #8 0x08091e86 in __ast_pbx_run (c=0x9e9cb80) at pbx.c:2250 #9 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #10 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #11 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 3 (process 30070): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x00fa9e27 in pthread_mutex_lock () from /lib/tls/libpthread.so.0 #4 0x080668f5 in ast_read (chan=0x2f6a9e) at channel.c:1945 #5 0x08069d98 in ast_channel_bridge (c0=0xb015fea0, c1=0xa0b5c88, config=0xb4e937a0, fo=0xb4e92c40, rc=0xb4e92c44) at channel.c:3399 #6 0x00548f65 in ast_bridge_call (chan=0xb015fea0, peer=0xa0b5c88, config=0xb4e937a0) at res_features.c:1365 #7 0x005a40ba in dial_exec_full (chan=0xb015fea0, data=Variable data is not available. ) at app_dial.c:1633 #8 0x005a6a33 in dial_exec (chan=0xfffc, data=0x348ff4) at app_dial.c:1680 #9 0x08090bad in pbx_extension_helper (c=0xb015fea0, con=Variable con is not available. ) at pbx.c:574 #10 0x08091e86 in __ast_pbx_run (c=0xb015fea0) at pbx.c:2250 #11 0x08093a2c in pbx_thread (data=0xfffc) at pbx.c:2537 #12 0x00fa83cc in start_thread () from /lib/tls/libpthread.so.0 #13 0x002e9c3e in clone () from /lib/tls/libc.so.6 Thread 2 (process 21752): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002f6a9e in __lll_mutex_lock_wait () from /lib/tls/libc.so.6 #2 0x0028800b in _L_mutex_lock_3800 () from /lib/tls/libc.so.6 #3 0x0028bb61 in strcasecmp () from
[asterisk-users] is there a way
hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because we
were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had to
use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If the
group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
___
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Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
On 21:28, Fri 10 Oct 08, Wayne wrote: Thanks both, The only thing I have a little concern over is that 1.6 is that its still a development release (if I understand things correctly). No, 1.6.0 has been released. This is indeed the first public 'final' release of the 1.6 series. But it's not in beta or release-candidate anymore. Basically, it's the latest and greatest version that should be stable. Stability is the main thing for me (its only a very small set up) but there are no technical people around if something were to go wrong through the day. You do know it's just another daemon an a linux box right ? If you cant afford downtime you should not bet on one server, but make every part of your network redundant. That means at least: connectivity power hardware locations backups all the other stuff I forgot -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Help need for debuging the core file.
On Friday 10 October 2008 15:42:34 gary wrote: I am running asterisk 1.2.27 and it dead today. The following is the backtrace of core file. Can anybody help me to identify what is the possible cause of crash? It seems the mysql connection causing problem in Thread 2. But I can not tell what exactly happened. This asterisk is using as ACD for over hundred agents. snip Thread 1 (process 30108): #0 0x00fc17a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x002487a5 in raise () from /lib/tls/libc.so.6 #2 0x0024a209 in abort () from /lib/tls/libc.so.6 #3 0x0027ca1a in __libc_message () from /lib/tls/libc.so.6 #4 0x002834c0 in _int_free () from /lib/tls/libc.so.6 #5 0x0028363a in free () from /lib/tls/libc.so.6 snip Please read doc/valgrind.txt. -- Tilghman ___ -- 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] Configuring Bandwidth.com SIP trunks to prevent one-way audio
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because
we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had to
use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If
the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
http://lists.digium.com/mailman/listinfo/asterisk-users
Is your Asterisk box on a public IP or behind a NAT/Firewall?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
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Re: [asterisk-users] Be aware of callcheap.com and Mike Low - It is scam
I also thought about it. Maybe I should not have posted it here. But I know he is actively searching for another company. Just don't want any other provider to suffer. Zeeshan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID service and the ethernet stucking
Hi All; We added the callerid service on our telephone line, once that done, now when we call to the Asterisk PBX or we need to place outside call via the digium (zaptel channel), the PBX got a problem in the network, and we become not able to reach it, this stay for a while of time (about 5 min) and then it come back reachable. I did not do any thing when the callerid service added by the telecom service provider, and I am surprised why this callerid service effect on the ethernet port? Did any one face this problem? My asterisk version: 1.4.19.2 My zaptel version: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10.1 Zaptel Echo Canceller: MG2 INFO-xpp: FEATURE: with sync_tick() from ZAPTEL In the /var/log/asterisk/messages, I did not find any message that help (warning or error). Any advise? Did any one face such problem? The PBX located in KSA. Regards Bilal ___ -- 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] Got event 17 (Polarity Reversal)...
Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Jim Tzafrir Cohen wrote: On Fri, Oct 10, 2008 at 12:57:27PM -0400, Jim Duda wrote: Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. grep polarity /etc/asteirsk/chan_dahdi.conf See the documentation for answeronpolarityswitch and hanguponpolarityswitch . ___ -- 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] is there a way
Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy What does your sip.conf look like?The only way I could see this happening would be if the IP's or Identities were somehow getting crossed up. Do your phones have static IP's or are they using DHCP? -Brent ___ -- 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] Be aware of callcheap.com and Mike Low - It is scam
On Fri, Oct 10, 2008 at 5:43 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I also thought about it. Maybe I should not have posted it here. But I know he is actively searching for another company. Just don't want any other provider to suffer. Zeeshan Again, your motives are admirable in my opinion, execution was flawed. If I were you though, I would post from a hushmail.com with no specific details that could tie the posting to you directly, just enough to put out the Heads Up I personally would also CC the person in question so they are aware of the posting and can reply. Maybe even incriminate themselves if you are lucky. The key is to keep your identity secret while bringing attention to the matter, if you feel you must, and give them a chance to defend themselves (or incriminate). The other thing is that your post really belongs on the -biz list and would probably have a better chance of alerting someone who may fall victim to a scammer. There is so much competition, that I assume it is quite easy to fall for a deal from someone that sounds sincere, to make a few dollars. I cannot really see setting up accounts with live trunks that are not on a prepaid system unless you have been dealing with the person/organization for quite some time, or you pull their business credit report and hold them personally responsible as well as their corporation through a legally binding contract. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] is there a way
On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson [EMAIL PROTECTED] wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy What does your sip.conf look like?The only way I could see this happening would be if the IP's or Identities were somehow getting crossed up. Do your phones have static IP's or are they using DHCP? -Brent I assume that he just has analog in his room and a basic hotel phone If they are SIP you stand a chance of figuring out without using social engineering, also if they have not separated the room net access from the PBX on the LAN. I have dualboot and use a very powerful free program put out by 3com called 3com network supervisor, the name has changed I think, but you can either search google or 3com and find the newest software. I am sure there is a Linux tool that does the same, just never bothered to find it since it is easy enough and free to dualboot and use the 3com software. It will go out and ping all the addresses you specify or would be included in your DHCP assigned subnet. It then tries to resolve hostnames, OS, services, and the like and give you a nice graphical map. A very good reason not to plug a laptop with open services and fileshares or whatever into a hotel network jack, or wifi. You will be shocked what you can find ~8-9PM in a large and full business type hotel. So once you map the IPs, look for something unusual or usual switches, routers, and hotel servers usually occupy the lower end of the IP pool. I have had totally open access to the hotels cisco switches and APs because they were never setup with passwords or used defaults. If you find a box that is running Linux, try the web interface and see if it identifies itself, like most flawed boxen do. So typing it's IP into a browser with http://IP or https://ip might tell you exactly what it is. Say it is a SwitchVox box https://ip/admin should tell you right way. Other devices that just pop up a login box will also tell you what the system is as I am sure you have seen with certain network devices, APs are a prime example. If you find that you may have identified an Asterisk box, try setting up a softphone and run wireshark while you register with your room number as the user and password. Many times, you will get logged in, because of poor implementation. If not but you get something back other than a timout, you can look at the SIP headers and try to determine from there. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because
we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as backup.
;This is where it breaks. I tried to make it so there can't be more than 2
calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had
to use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If
the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
http://lists.digium.com/mailman/listinfo/asterisk-users
Is your Asterisk box on a public IP or behind a NAT/Firewall?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Configuring Bandwidth.com SIP trunks to prevent one-way audio
You need to configure your box for nat settings, externip and other settings
in sip.conf and set nat=yes instead of nat=no.
One way audio is almost always a NAT issue and those are two glaring things
that would cause problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and
someone tries to dial out, they cause another call to get one-way audio (the
caller hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband because
we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix it
added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as
backup.
;This is where it breaks. I tried to make it so there can't be more than
2 calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had
to use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If
the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
http://lists.digium.com/mailman/listinfo/asterisk-users
Is your Asterisk box on a public IP or behind a NAT/Firewall?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
-- 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
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
--
Re: [asterisk-users] Menu for call forwarding or voicemail
Any reason not to ring both at once? exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],20) -Darren That would also work but what if my sip/101 device (softphone) isn't connected. Currently if my softphone is not connected then the line will go straight to voicemail. If I remove the voicemail to implement your rule then it will error out since the phone isn't connected. ___ -- 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] Budge Tones pick up wrong calls
First of all, are the handsets using distinct SIP peers? Are they set up statically or to register? Secondly, unless you are using an Ethernet hub, SIP signaling data destined for one phone should not go to another. Paul Douglas Franklin wrote: We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For example, 2501 might respond to the calls for 2518. After a reboot, it might decide to respond to 2501 as it should. Or it might respond to 2536. The phone it responds for will not respond. I don't know whether to look in the settings on the phone or in an Asterisk setting, and what setting to check in either place. Has anyone seen this behavior before? --Paul -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] is there a way
no i'm a guest at the bestwestern On Oct 10, 2008, at 1:55 PM, Brent Davidson wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy What does your sip.conf look like?The only way I could see this happening would be if the IP's or Identities were somehow getting crossed up. Do your phones have static IP's or are they using DHCP? -Brent ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] is there a way
steve; thanks a lot mike On Oct 10, 2008, at 2:20 PM, Steve Totaro wrote: On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson [EMAIL PROTECTED] wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy What does your sip.conf look like?The only way I could see this happening would be if the IP's or Identities were somehow getting crossed up. Do your phones have static IP's or are they using DHCP? -Brent I assume that he just has analog in his room and a basic hotel phone If they are SIP you stand a chance of figuring out without using social engineering, also if they have not separated the room net access from the PBX on the LAN. I have dualboot and use a very powerful free program put out by 3com called 3com network supervisor, the name has changed I think, but you can either search google or 3com and find the newest software. I am sure there is a Linux tool that does the same, just never bothered to find it since it is easy enough and free to dualboot and use the 3com software. It will go out and ping all the addresses you specify or would be included in your DHCP assigned subnet. It then tries to resolve hostnames, OS, services, and the like and give you a nice graphical map. A very good reason not to plug a laptop with open services and fileshares or whatever into a hotel network jack, or wifi. You will be shocked what you can find ~8-9PM in a large and full business type hotel. So once you map the IPs, look for something unusual or usual switches, routers, and hotel servers usually occupy the lower end of the IP pool. I have had totally open access to the hotels cisco switches and APs because they were never setup with passwords or used defaults. If you find a box that is running Linux, try the web interface and see if it identifies itself, like most flawed boxen do. So typing it's IP into a browser with http://IP or https://ip might tell you exactly what it is. Say it is a SwitchVox box https://ip/admin should tell you right way. Other devices that just pop up a login box will also tell you what the system is as I am sure you have seen with certain network devices, APs are a prime example. If you find that you may have identified an Asterisk box, try setting up a softphone and run wireshark while you register with your room number as the user and password. Many times, you will get logged in, because of poor implementation. If not but you get something back other than a timout, you can look at the SIP headers and try to determine from there. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] Configuring Bandwidth.com SIP trunks to prevent one-way audio
externip messes up DTMF detection, and by messes up I mean it doesn't detect
it at all. Setting nat=yes or nat=no didn't make a difference either.
When the trunks are in use, the calls are fine, no dropped audio. It only
happens when a 3rd call is made and there's no trunk available.
Thanks :)
On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
You need to configure your box for nat settings, externip and other
settings in sip.conf and set nat=yes instead of nat=no.
One way audio is almost always a NAT issue and those are two glaring things
that would cause problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and
someone tries to dial out, they cause another call to get one-way audio
(the
caller hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband
because we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as
backup.
;This is where it breaks. I tried to make it so there can't be more than
2 calls on SIP channels at once.
;Since it counts the phone as a channel, and adds it to the group, I had
to use 4.
[internalphones]
exten = _1NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _1NXXNXX,n,GotoIf($[${GROUP_COUNT(SIPGROUP)} = 4]?100) ;If
the group has 2 or more calls, do not dial.
exten = _1NXXNXX,n,NoOp(1NCount = ${GROUP_COUNT(SIPGROUP)})
exten =
_1NXXNXX,n,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_1,trunk_2)
exten = _1NXXNXX,100,Playback(all-circuits-busy-now)
exten = _1NXXNXX,101,congestion()
exten = _1NXXNXX,102,busy()
;This is where incoming calls go to if I'm awake.
[DID_trunk_2_timeinterval_Awake]
exten = _NXXNXX,1,Set(GROUP()=SIPGROUP)
exten = _NXXNXX,n,NoOp(Open Count=${GROUP_COUNT(SIPGROUP)})
exten = _NXXNXX,n,Set(CALLERID(num)=1${CALLERID(num)})
exten = _NXXNXX,n,Goto(voicemenu-custom-1|s|1)
Thanks.
http://lists.digium.com/mailman/listinfo/asterisk-users
Is your Asterisk box on a public IP or behind a NAT/Firewall?
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
-- 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] Configuring Bandwidth.com SIP trunks to prevent one-way audio
Oh, I thought you had logic to count the calls on the trunk. You should
limit each trunk to one call. This is the primary reason besides the email
that basically said that customer support structure has been changed and
anything beyond the Demarc would not be supported, I the Demarc is simply
their boxen, so unless it is on their side, you will not get any helpful
support from Bandwidth, plus they CCed over 500 people by address instead of
setting up a group.
http://www.bandwidth.com/content/support/?page=standardSupport
I am with Junction and while a trunk is not unlimited as far as price for
usage, the amount of trunks is unlimited (or at least as unlimited as it can
be since nothing is really unlimited). They asked that I try not to go over
one call per second for any real duration, and that I not hammer one LATA do
to limited interconnects.
The other thing was Junctions was very easy to sign up with, great support,
and configuration was a breeze.
As for Bandwidth, I think they are solid but due to recent changes and the
fact that you must pay per channel, as well as the setup process, I decided
they were not for me.
I will take a second look at your sip.conf and extensions.conf later to see
if something jumps out at me. I suspect since you are setting up two
separate trunks with Bandwidth, you need to limit each trunk to one call,
rather than two.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 9:47 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
externip messes up DTMF detection, and by messes up I mean it doesn't
detect it at all. Setting nat=yes or nat=no didn't make a difference either.
When the trunks are in use, the calls are fine, no dropped audio. It only
happens when a 3rd call is made and there's no trunk available.
Thanks :)
On Fri, Oct 10, 2008 at 7:09 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
You need to configure your box for nat settings, externip and other
settings in sip.conf and set nat=yes instead of nat=no.
One way audio is almost always a NAT issue and those are two glaring
things that would cause problems.
Thanks,
Steve Totaro
On Fri, Oct 10, 2008 at 6:32 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hi Steve,
It's behind a NAT/Firewall but SIP translation is enabled and removing it
from behind the firewall did nothing, it still dropped calls. The calls
connect and everything works, but it dies when all trunks are in use and
someone else tries to call out. It seems like even though both channels are
in use, it tries to connect to the 2nd trunk and thus kills the audio.
Nothing strange came up in Wireshark or the firewall logs.
Thanks.
On Fri, Oct 10, 2008 at 5:40 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Oct 10, 2008 at 5:17 PM, Kurt Knudsen [EMAIL PROTECTED]wrote:
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and
someone tries to dial out, they cause another call to get one-way audio
(the
caller hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that
tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
currently using, or attempting to use, groups to solve this problem, but
sometimes it works, sometimes it doesn't. It breaks when a call goes out
on
a Queue, because it seems to add each phone to the group, which breaks my
GotoIf() statement. Here's some relevant information:
Users.conf (added by Asterisk-GUI)
[trunk_2]
provider = Bandwidth (SIP) ; GUI metadata
context = DID_trunk_2
hasexten = no
hasiax = no
hassip = yes
host = 216.82.224.202
registeriax = no
registersip = no
usecallerid = yes
nat = no ;Testing
trunkname = Bandwidth.com (Sip) ; GUI metadata
username =
secret =
disallow = all
allow = ulaw,alaw,g726
sip.conf
[general]
context = frombandwidth
;other variables, etc.
;Added according to Bandwidth.com's wiki entry. Changed to inband
because we were having DTMF issues.
[bandwidth.com_inbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=inband
canreinvite=no
reinvite=no
context=frombandwidth
nat=no
[bandwidth.com_outbound]
host=216.82.224.202
port=5060
type=peer
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=no
fromuser=11234567890
extensions.conf
[globals]
;…irrelevant stuff
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
OUT_2 = SIP/bandwidth.com_outbound
;Took out the Set(GROUP()) because I moved it elsewhere to try and fix
it added all the phones when Asterisk calls agents on a Queue.
[frombandwidth]
;exten = _+1.,1,Set(GROUP()=SIPGROUP)
exten = _+1.,1,NoOp(FromCount=${GROUP_COUNT(SIPGROUP)})
exten = _+1.,n,Set(DID=${EXTEN:2})
exten = _+1.,n,Set(CALLERID(num)=${CALLERID(num):2})
exten = _+1.,n,Goto(DID_trunk_2,${DID},1)
;What we use to dialout. Try SIP trunks first, then Dahdi trunk as
backup.
;This is where it breaks. I
[asterisk-users] Mitel 5220 firmware
Hi, Just wondering if anyone here has the latest SIP firmware for the Mitel 5220. I have a few of these phones with the latest firmware (6.0.0.19) that is on the Mitel site. However I believe the latest version is 7.2 and they only make it available to partners and resellers. Anyone out there with version 7.2? Thanks, Bro. ___ -- 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] is there a way
nmap for scanning and identification. cross platform and even a nice gui for windows. Eric On Fri, Oct 10, 2008 at 3:20 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson [EMAIL PROTECTED] wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy What does your sip.conf look like?The only way I could see this happening would be if the IP's or Identities were somehow getting crossed up. Do your phones have static IP's or are they using DHCP? -Brent I assume that he just has analog in his room and a basic hotel phone If they are SIP you stand a chance of figuring out without using social engineering, also if they have not separated the room net access from the PBX on the LAN. I have dualboot and use a very powerful free program put out by 3com called 3com network supervisor, the name has changed I think, but you can either search google or 3com and find the newest software. I am sure there is a Linux tool that does the same, just never bothered to find it since it is easy enough and free to dualboot and use the 3com software. It will go out and ping all the addresses you specify or would be included in your DHCP assigned subnet. It then tries to resolve hostnames, OS, services, and the like and give you a nice graphical map. A very good reason not to plug a laptop with open services and fileshares or whatever into a hotel network jack, or wifi. You will be shocked what you can find ~8-9PM in a large and full business type hotel. So once you map the IPs, look for something unusual or usual switches, routers, and hotel servers usually occupy the lower end of the IP pool. I have had totally open access to the hotels cisco switches and APs because they were never setup with passwords or used defaults. If you find a box that is running Linux, try the web interface and see if it identifies itself, like most flawed boxen do. So typing it's IP into a browser with http://IP or https://ip might tell you exactly what it is. Say it is a SwitchVox box https://ip/admin should tell you right way. Other devices that just pop up a login box will also tell you what the system is as I am sure you have seen with certain network devices, APs are a prime example. If you find that you may have identified an Asterisk box, try setting up a softphone and run wireshark while you register with your room number as the user and password. Many times, you will get logged in, because of poor implementation. If not but you get something back other than a timout, you can look at the SIP headers and try to determine from there. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Budge Tones pick up wrong calls
Are you using NAT? On Fri, Oct 10, 2008 at 4:24 PM, Paul Douglas Franklin [EMAIL PROTECTED] wrote: We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For example, 2501 might respond to the calls for 2518. After a reboot, it might decide to respond to 2501 as it should. Or it might respond to 2536. The phone it responds for will not respond. I don't know whether to look in the settings on the phone or in an Asterisk setting, and what setting to check in either place. Has anyone seen this behavior before? ___ -- 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] is there a way
I will look into that when I get my Acer Aspire One running FC8, it came with windows XP and I got the 1gig ram, 120gig HD. I am following threads on howto but nobody has a definitive guide yet, that allows the embedded webcam and the NIC to work properly. Maybe (probably) my USB Alpha AWUS036H with upgradable antenna will probably be much better than the stock onboard NIC. Plus it supports packet injections which is nice. Thanks, Steve Totaro My only wish is that Linux had a facility like XP to bridge NICs without running all sorts of commands for brctl. Just a GUI like XP. Last time I setup a bridge in Linux, I had to change many kernel options and rebuild the entire kernel to get bridging working properly. With XP, you just select the NICS, right click and select add to bridge. For linux, I find that running firestarter, ICS/Firewall is fine, my end game is to get all of my traffic to go over an OpenVPN tunnel at my colo which is the default gateway over OpenVPN. Windows seems to have the easiest method of getting this done. Thanks, Steve Totaro On Fri, Oct 10, 2008 at 10:33 PM, Eric Fort [EMAIL PROTECTED] wrote: nmap for scanning and identification. cross platform and even a nice gui for windows. Eric On Fri, Oct 10, 2008 at 3:20 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Oct 10, 2008 at 5:55 PM, Brent Davidson [EMAIL PROTECTED] wrote: Babcock, Michael Alex wrote: hey; i'm at best western and am curious is there a way i could find out if our best western, with out asking, is using asterisk? oh and petsmart i think is using asterisk they have alason voice for there main voicem enu. mike thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy What does your sip.conf look like?The only way I could see this happening would be if the IP's or Identities were somehow getting crossed up. Do your phones have static IP's or are they using DHCP? -Brent I assume that he just has analog in his room and a basic hotel phone If they are SIP you stand a chance of figuring out without using social engineering, also if they have not separated the room net access from the PBX on the LAN. I have dualboot and use a very powerful free program put out by 3com called 3com network supervisor, the name has changed I think, but you can either search google or 3com and find the newest software. I am sure there is a Linux tool that does the same, just never bothered to find it since it is easy enough and free to dualboot and use the 3com software. It will go out and ping all the addresses you specify or would be included in your DHCP assigned subnet. It then tries to resolve hostnames, OS, services, and the like and give you a nice graphical map. A very good reason not to plug a laptop with open services and fileshares or whatever into a hotel network jack, or wifi. You will be shocked what you can find ~8-9PM in a large and full business type hotel. So once you map the IPs, look for something unusual or usual switches, routers, and hotel servers usually occupy the lower end of the IP pool. I have had totally open access to the hotels cisco switches and APs because they were never setup with passwords or used defaults. If you find a box that is running Linux, try the web interface and see if it identifies itself, like most flawed boxen do. So typing it's IP into a browser with http://IP or https://ip might tell you exactly what it is. Say it is a SwitchVox box https://ip/admin should tell you right way. Other devices that just pop up a login box will also tell you what the system is as I am sure you have seen with certain network devices, APs are a prime example. If you find that you may have identified an Asterisk box, try setting up a softphone and run wireshark while you register with your room number as the user and password. Many times, you will get logged in, because of poor implementation. If not but you get something back other than a timout, you can look at the SIP headers and try to determine from there. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing
Re: [asterisk-users] Got event 17 (Polarity Reversal)...
On Fri, Oct 10, 2008 at 05:51:29PM -0400, Jim Duda wrote: Tzafrir, Thanks for the tip. I'm researching answeronpoliaryswitch. I suspect this will solve my issue. I never would have know to look for this. Thanks much! You made my day :-) Hmm... I might have misled you. By default Asterisk ignores all polarity events. Using the polarity events can be a useful feature, but I suspect that it is not the cause of your original problem. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] is there a way
On Fri, Oct 10, 2008 at 07:33:45PM -0700, Eric Fort wrote: nmap for scanning and identification. cross platform and even a nice gui for windows. What nmap does is called fingerprinting. it mostly uses the fact that when faced with normal behaviours, most stacks behave the same. But when faced with non-standard behaviour, different stacks would react differently. I figure you're actually looking for a more higher-level fingerprinting. 'sip fingerprinting' gives some results. I never tested any. Looking at a SIP trace for the name of the remote agent (does it happen to be Asterisk) is a good start. But then again, you can try your own. * Is there a voicemail menu? If so, what is its structure? * Are there any conference rooms suggested? Look familiar? * The default installation of FreePBX will respond to a misdialed call with a friendly 'Your call cannot be connected' IIRC. Though this can be configured off. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
