[Asterisk-Users] ISDN phone Hold-Problem connected to QuadBRI/Zap
Folks, (sorry for overlong lines) I have recently configured one port on my QuadBRI card to work in NT mode with NET signalling configured so that I can use an ISDN telephone on it. I have set up a separate group in zapata.conf and can call the phone and place calls from it like a charm. No problems at all. Problems came up when trying to hold a call and get it back. I turned on pri debug span x and this is what I get: --- Placing a call to the ISDN phone -- -- Executing SetVar(SIP/11-e108, ALERT_INFO=Bellcore-Stutter) in new stack -- Executing Dial(SIP/11-e108, Zap/g8/18|15) in new stack -- Making new call for cr 132 Protocol Discriminator: Q.931 (8) len=38 Call Ref: len= 1 (reference 4/0x4) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] [28 0d 53 61 73 63 68 61 20 50 6f 6c 6c 6f 6b] Display (len=13) [ xx ] [6c 04 41 80 31 31] Calling Number (len= 6) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '11' ] [70 03 c1 31 38] Called Number (len= 5) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '18' ] -- Called g8/18 Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 132/0x84) (Terminator) Message type: CALL PROCEEDING (2) -- Zap/10-1 is making progress passing it to SIP/11-e108 Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 132/0x84) (Terminator) Message type: ALERTING (1) -- Zap/10-1 is ringing Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 132/0x84) (Terminator) Message type: CONNECT (7) Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 4/0x4) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/10-1 answered SIP/11-e108 - ISDN phone putting the caller on hold. Caller can hear MoH - Actually ISDN phone is using the New call button Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 132/0x84) (Terminator) Message type: HOLD (36) -- Started music on hold, class 'default', on SIP/11-e108 Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 4/0x4) (Originator) Message type: HOLD ACKNOWLEDGE (40) -- Hungup 'Zap/10-1' == Spawn extension (internal-in, 18, 2) exited non-zero on 'Onhold/SIP/11-e108ZOMBIE' Protocol Discriminator: Q.931 (8) len=19 Call Ref: len= 1 (reference 14/0xE) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [6c 04 01 80 31 38] Calling Number (len= 6) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '18' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) -- Making new call for cr 14 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 125 (cs0, High-layer Compatibility) Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 1 (reference 142/0x8E) (Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 01 8a] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B2 channel ] [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Starting simple switch on 'Zap/11-1' -- Accepting overlap call from '18' to 'unspecified' on channel 0/2, span 4 ^^^ This is maybe because the caller pressed the new call button -- I did not try to dial a number on the isdn phone just -- tried to resume the 1st call Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 14/0xE) (Originator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/2, span 4 got
Re: [Asterisk-Users] CAC Access Bank Manual
Hi, The asterisk configuration and the channel bank configuration are both set to esf and b8zs. Howerver I am still getting the framing Error Red and blinking. zttool shows there are no alarms. According to the manual, Framing Error (Red and Blinking )means Network T1 is out of frame (received signal cannot be framed to ESF or D5 as configured by T1 Option switch 4) I tried with both DIP switch on and off, but no help. Any ideas ? Is my card or channel bank bad ? On Thursday 17 March 2005 19:12, Jerry wrote: Carrier Access generally have all of their manuals available for download. You just have to request a free login. they also provide excellent dialin support - also free. If your framing LED is blinking I would double check that both ends of your span are set for ESF. zttool is the tool for working on the cards. On Mar 17, 2005, at 4:40 AM, Vicky Shrestha wrote: Hi, Does anyone have Carrier Access Corporation (CAC) Access Bank I Manual ? Could you please email it to me off list ? We have a FXS channel bank and the framing Error Led is blinking and I have no clue on what could be the problem . Is there command line utilities available in Linux to Troubleshoot T1 connection using Zaptel drivers ? /etc/zaptel.conf = span=1,1,0,esf,b8zs #span=1,1,0,esf,ami #span=1,1,0,d4,b8zs #span=1,1,0,d4,ami #em=1-24 fxols=1-24 loadzone=us defaultzone=us == /etc/asterisk/zapata.conf = [channels] language=us context=default signalling=fxo_ls ;usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes ;threewaycalling=yes transfer=yes ;cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group = 1 channel = 1-24 === dmesg output = Zapata Telephony Interface Registered on major 196 Found TE410P at base address dfcdff80, remapped to d0e23f80 TE410P version c01a009b, burst ON FALC version: 0005, Board ID: 00 Reg 0: 0x0e3c6800 Reg 1: 0x0e3c6000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source == -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, therethick!
Brian, I am reluctant to post against you. However, for my previous 2 emails simply based on facts you as a third person have over-responded, with no good reasons, in the exact way that you commented me. YOU proved yourself, not me. Are you that guy the phone number is associated with? That'll explain the exact reason you do this to me. People can read and understand what I have been doing here. There will be NO more from me. V. On Fri, 18 Mar 2005 01:49:38 -0500, Brian Capouch [EMAIL PROTECTED] wrote: Vincent wrote: Hi all, You don't want to be fooled by - -. This guy has NO business ethic. When He refused to realize a business deal in which I agreed to pay for his coding help for me, will he personally pay for the hosting of the list? More interestingly, he mentioned in the list that he lives in in Timbuktu, Ontario while he told me that he lives in Asheville, North Carolina but home number is a Hendersonville, NC phone number. I was just updated that according to the phone company records that is not the name of the person the phone number is associated with. Vincent I'm sorry to report that your bitter, petty, and unethical attempt to gain vengance against this list member is totally transparent. It will have no effect on anyone's opinion of the target of your rant, but great effect on everyone's opinion of you. You abused the mailing list, you took a personal gripe public in a vulgar way, and you proved your own cluelessness by not getting his joke about being from Timbuktu. Next time you should count to, um, infinity before posting crap like this to the list. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
Vicky Shrestha wrote: The asterisk configuration and the channel bank configuration are both set to esf and b8zs. Howerver I am still getting the framing Error Red and blinking. zttool shows there are no alarms. According to the manual, Framing Error (Red and Blinking )means Network T1 is out of frame (received signal cannot be framed to ESF or D5 as configured by T1 Option switch 4) I tried with both DIP switch on and off, but no help. Any ideas ? Is my card or channel bank bad ? Probably not. More likely your CAC ABI is set up to run in TR08 mode which is incompatible with standard T1 framing. Check the LIU board (the Line Interface Board -- as opposed to the FXO or FXS cards) and look at the PROM. It usually will be marked TR08. If that is the case you will need to order a D4/ESF upgrade kit. If you have a 1.x revision TR08 chip, you will need P/N 750-0018. If you have a 3.x revision TR08 chip, you will need P/N 750-0019. Good luck! -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
On Fri, 2005-03-18 at 14:05 +0545, Vicky Shrestha wrote: Hi, The asterisk configuration and the channel bank configuration are both set to esf and b8zs. Howerver I am still getting the framing Error Red and blinking. zttool shows there are no alarms. According to the manual, Framing Error (Red and Blinking )means Network T1 is out of frame (received signal cannot be framed to ESF or D5 as configured by T1 Option switch 4) I tried with both DIP switch on and off, but no help. /etc/zaptel.conf = span=1,1,0,esf,b8zs Do you have the CAC set to provide timing to the line? If not, you need to set your timing to 0 here so the TE410P card will provide timing. Also, as a precaution, It is helpful to power cycle the machine when you change timing. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue log analyser?
any chance of seeing some code soon? On Jan 20, 2005, at 12:17, Ben Merrills wrote: I've not released the source yet, I asked last week on the mailing list for people to send me over some example queue_logs, because so far I've only been able to test the software against my own. I have however made a lot of changes to it since last I posted about it. Template engine has been improved Allows for recursion of a directory of templates Allows for different output directories (so you can do a daily, weekly and monthly all from the same set of templates say) And quite a few other bits As soon as I get some sample data that people don't mind the results being posted for then I can show it off a bit more. Hope to get some sample data soon, Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of João Amaro Sent: 20 January 2005 11:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] queue log analyser? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ben Merrills wrote: | There's a few (open source/free) ones in development. I myself am | developing one of them. | | Ben | Hi. Why not join all the project in just one ? Actually which queue log analyzers projects are beeing developed ? Check the mail from Ben Merrills sent to the list 14-10-2004 15:10. I don't know if he releases the source code, but, from the screenshots it seems to be a good one. Jo?o Amaro - -- Begin Mail | I've been doing some work on a queue log analyser for a while now, | getting the basics in place, an example of which you can find at | the URL below. However, just wondering what information people | think is most useful in a log analyser? | | At present it includes the following features: | | # Time periods - specify a period of days from the log which you | want to generate statistics for (e.g. only the last 14 days) # | Templating - allows the stats to be inserted into any html/text | template using specific tags to insert stats. This means you could | create a number of templates and execute the analyser against them | to give different information on different pages (quite flexible). | # Specify start and end dates - similar to the first feature, | except you can specify a tight period from your log, not just the | last x number of days # Channels/Agents to names - simple text file | allows you to specify a name, agent number and a channel - e.g. | Ben, Agent/1, Sip/ben. This is then used in the output # instead | of raw data # JPG graphs - includes a custom class to generate line | graphs of information (e.g. hourly call volumes etc) | | What I want to know though is, what output people would like. At | the moment there is an overview of all queues, which includes: | | Total Calls, total connected calls, total abandoned calls, calls | abandoned within x seconds, calls exited with key press, Average | hold time, max hold time, average talk time | | Agent overview includes: Calls taken, Average talk time | | Graph of call volume per hour of the day Graph of call volume per | day (over the period specified) | | Runs under windows (.NET or mono required) or any other OS that | support .NET/mono (Linux, Mac, BSD etc) | | http://muad.xdev.net/Projects/qig/sample.html | | | Not really done anything like this before, so as much input as | possible would be appreciated. | | Cheers, | | Ben -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFB75EfJUm/Bor63CERAv/UAKCwiYZ96RLqX0m7Ks9eL1f7iG4IDQCcCWvK gafg+vLAgQpjl75Hp5y8tug= =PwR8 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail, busy does not work?
hallo, i tried to setup my extentions,conf like this but it never jumps to the busy part (102) asterisk always plays the unavail msg, also when i am connected to another iax channel (conferece room) and no more channel on my client is available. could sombody give me a hint what could be wrong? thanks , alex snd*CLI -- Accepting AUTHENTICATED call from 81.135.10.114, requested format = 1024, actual format = 1024 -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, IAX2/atucek|10|Ttr) in new stack -- Called atucek Mar 18 09:54:28 WARNING[21135]: chan_iax2.c:5549 socket_read: Call rejected by 81.135.10.114: Too many calls, we're busy! -- Hungup 'IAX2/atucek/10' == No one is available to answer at this time -- Executing NoOp(IAX2/[EMAIL PROTECTED]/3, NOANSWER) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/3, u) in new stack -- Playing 'voicemail/default//unavail' (language 'en') -- Playing 'vm-intro' (language 'en') [default] exten = ,1,Dial(IAX2/atucek,10,Ttr) exten = ,2,Voicemail(u) exten = ,102,Voicemail(b) exten = ,103,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P Not loading Drivers
I am trying to get the drivers working with this device with 4 fxo modules on it. I do a modprobe zaptel and no errors appear. But when I do modprobe wctdm the following errors appear: Notice: Configuration file is /etc/zaptel.conf line 4: Cannot get number of tones chanel 1 line 4: Cannot init tones chanel 1 line 4: Cannot get number of tones chanel 2 line 4: Cannot init tones chanel 2 line 4: Cannot get number of tones chanel 3 line 4: Cannot init tones chanel 3 line 4: Cannot get number of tones chanel 4 line 4: Cannot init tones chanel 4 8 error(s) detected /lib/modules/2.4.22-2f/misc/wctdm.o: post-install wctdm failed /lib/modules/2.4.22-2f/misc/wctdm.o: insmod wctdm failed I have googled the mailing lists and have not been able to find anything on these errors. My config of zaptel.conf fxsks=1-4 loadzone=au defaultzone=au channels=1-4 zapata.conf [channels] language=en ; XTDM20B Port #1,2 plugged into PSTN ; context=from-1800 signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no group=1 channel=1 context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=3 channel=2-3 context=from-fax signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=3 channel=4 Has anyone seen this problem before? Does anyone know a cure? Any help is greatly appreciated. Regards, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules)
I'm afraid I am at a loss. If the three files, app_cbmysql.c, app_meetme2.c and Makefile all exist in ../apps then a patch -p1 from the ../asterisk directory should work. The -p1 tells patch to ignore the first directory in the path to the file in the patch, -p2 ignores two directories. Another option is to just edit the apps-meetme-cbmysql.txt and split it into three patchs and apply them one at a time. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Thursday, March 17, 2005 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules) It doesn't matter if I run it from the apps directory or the asterisk directory I get the same response. This is getting frustrating. - Original Message - From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 18, 2005 12:17 AM Subject: RE: [Asterisk-Users] ANNOUNCEMENT: Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules) I should have read a little closer- [EMAIL PROTECTED] apps]# patch -p1 /var/build_aah/asterisk_src/asterisk/apps/apps-meetme-cbmysql.txt If you run patch from within the apps directory, you will need to use -p2. Or just cd .. and use the same command as above. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail configuration
Hi all, Is it possible to modify the voicemail scenario (for example : group the digits by 2), add or erase some questions ? And if yes where i can see to do that ? thx in advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk+radius
hello pongco if you are talking about disconnecting a call session at his credit time. then you have to look at ast_channel-whentohangup kamran On Fri, 2005-03-18 at 14:10, Paul P. Pongco wrote: Hello, Im actually deciding if I will use asterisk+radius for AAA purposes or use logging directly to mysql and using Asterisk+RealTime to store SIP users to mysql also. Question is, what's the best way to disconnect a user, if for example, he runs out of credits. thanks. On Fri, 2005-03-18 at 02:33, izo wrote: set asterisk to log into database directly via there are mysql , postgresql and odbc drivers available. You dont need radius at all, for billing and accounting all u need is a frontend to database On Thu, 17 Mar 2005 12:29:34 -0500, Matt wrote: Oh this is sad.. I'm familiar with radius.. and was hoping to be able to use asterisk with freeradius to be able to do call accounting and billing.. so you're telling me this is now not a good idea? Am I better off (for now) parsing the csv report each month? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Three way calling with X-Lite / MeetMe
There is a crack available: http://www.regnow.com/softsell/nph-softsell.cgi?item=9054-12 You're suggesting eyeBeam ? Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom vs. Cisco IP Phones
Noah Miller wrote: My experience is that the Cisco and Polycom phones are both about in terms audio quality and useability. Neither one does exactly what I'd expect with respect to multiple lines. They both take a little extra setup in this regard, but you can read the wiki for that stuff. Snoms do exactly what I'd expect for a multiple line phone, are very easy to setup, but the audio quality and usability do not compare favorably with either Cisco or Polycom. If you've considered the Snom, you might also want to test a Zultys 4x4 or 4x5. I picked a 4x5 up off of ebay recently and have been pleasantly surprised by it. While I don't currently have a Polycom to compare it with, I would rank the audio quality equal to the Cisco's. It also just 'does the right thing' with multiple lines - only one registration, no hints needed. Can be configured through TFTP with both default and phone specific config files. Software updates are freely available from the Zultys website. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pattern matching in extensions.conf
Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ## 00 ## 20 ## 30 ## 40 ## 15 ## 35 ## 12 ## 44 Right now I've solved it by doing this: exten = _##[0234]0,1,HangUp exten = _##[13]5,1,HangUp exten = ##12,1,HangUp exten = ##44,1,HangUp The ## symbolises a fixed number, it's just censored away (and not important anyway) I was just wondering if there was a more intelligent approach, so this could be combined into one extension. Best regards, Mickey Binder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID problem
Sounds like your provider may be using different carriers for terminiation and some may not support CID as you send it. If it is handed off to a traditional IXC through your PSTN (LEC) connection then they will see whatever number is associated with the billing number (BAN). Your provider may be doing LCR and some routes may not work with CID. James On Thu, 17 Mar 2005 11:59:06 -0500, Oswaldo Arratia [EMAIL PROTECTED] wrote: Hi List I've been using Asterisk for quite some time with no major problems, but I've been facing this bug from the beginning and now I want to see if that is fixable. We have a provider who terminates our USA LD traffic and the problem comes when relaying the caller ID I send them from my Asterisk. Here is the weird thing, I send a call with valid caller ID info (areacode+number); my provider gets the call and routes it properly, the end receiver gets the call and does not see the caller ID I sent, they just get 'Unknown Number'. This remains true for some receivers, others using a different telephone company or cellular company do get the caller ID I sent. Examples: Cingular, Verizon do not show my caller ID info Nextel, T-mobile do show my caller ID info Is there something I am not following or not doing it industry standard? Thanks Oswaldo A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pattern matching in extensions.conf
What is 00 and other numbers? Are different destinations prefix ?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: vendredi 18 mars 2005 12:39 To: Asterisk maillist (asterisk-users@lists.digium.com) Subject: [Asterisk-Users] Pattern matching in extensions.conf Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ## 00 ## 20 ## 30 ## 40 ## 15 ## 35 ## 12 ## 44 Right now I've solved it by doing this: exten = _##[0234]0,1,HangUp exten = _##[13]5,1,HangUp exten = ##12,1,HangUp exten = ##44,1,HangUp The ## symbolises a fixed number, it's just censored away (and not important anyway) I was just wondering if there was a more intelligent approach, so this could be combined into one extension. Best regards, Mickey Binder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7940 convert to sip
Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar 18 15:12:30 AKrasavin xinetd[10068]: EXIT: tftp pid=10080 duration=0(sec) Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10082 from=192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10083]: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10083]: peer requests SEP000AF4BB7D59.cnf.xml, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10083]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: EXIT: tftp pid=10082 duration=0(sec) OS79XX.TXT and SEP000AF4BB7D59.cnf.xml exists. I'll be very thankful for any your help. -- WBR, Krasavin Andrey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Not loading Drivers
Inline... I am trying to get the drivers working with this device with 4 fxo modules on it. I do a modprobe zaptel and no errors appear. But when I do modprobe wctdm the following errors appear: Notice: Configuration file is /etc/zaptel.conf line 4: Cannot get number of tones chanel 1 line 4: Cannot init tones chanel 1 line 4: Cannot get number of tones chanel 2 line 4: Cannot init tones chanel 2 line 4: Cannot get number of tones chanel 3 line 4: Cannot init tones chanel 3 line 4: Cannot get number of tones chanel 4 line 4: Cannot init tones chanel 4 8 error(s) detected /lib/modules/2.4.22-2f/misc/wctdm.o: post-install wctdm failed /lib/modules/2.4.22-2f/misc/wctdm.o: insmod wctdm failed I have googled the mailing lists and have not been able to find anything on these errors. My config of zaptel.conf fxsks=1-4 loadzone=au defaultzone=au channels=1-4 Unless the above is a typo, channels=1-4 should not be in this file; only the first three lines zapata.conf [channels] language=en ; XTDM20B Port #1,2 plugged into PSTN ; context=from-1800 signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no group=1 channel=1 Not sure about the syntax of channel=1; I know channel=1 works though. context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=3 channel=2-3 context=from-fax signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=3 channel=4 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Pattern matching in extensions.conf
What is 00 and other numbers? Are different destinations prefix ?? Nope, it's just the last 2 digits of some 8 digit numbers that isn't supposed to be reachable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mickey Binder Sent: vendredi 18 mars 2005 12:39 To: Asterisk maillist (asterisk-users@lists.digium.com) Subject: [Asterisk-Users] Pattern matching in extensions.conf Hello fellow * users Hope this isn't a stupid question; I've done my research but could not find a proper answer. I have 8 different destinations which I want to match. The numbers are: ## 00 ## 20 ## 30 ## 40 ## 15 ## 35 ## 12 ## 44 Right now I've solved it by doing this: exten = _##[0234]0,1,HangUp exten = _##[13]5,1,HangUp exten = ##12,1,HangUp exten = ##44,1,HangUp The ## symbolises a fixed number, it's just censored away (and not important anyway) I was just wondering if there was a more intelligent approach, so this could be combined into one extension. Best regards, Mickey Binder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Group Ring after Timeout
Dear All, Would like to know what I should do to:: pickup call immediately and simultaneously Ring a Group, so that caller is listening to message whilst group phones are ringing and first one to pickup gets the call. Thanks :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to place calling rule contexts?
If I only want to give my sip users say local calling where do I put
that in the sip config?
I have the contexts setup.
[outbound-local]
exten = _NXX,1,Macro(dialout-default,${EXTEN})
exten = _NXXNXX,1,Macro(dialout-default,${EXTEN})
[outbound-tollfree]
exten = _1800NXX,1,Macro(dialout-default,${EXTEN})
exten = _1888NXX,1,Macro(dialout-default,${EXTEN})
exten = _1877NXX,1,Macro(dialout-default,${EXTEN})
exten = _1866NXX,1,Macro(dialout-default,${EXTEN})
[outbound-ld]
exten = _1NXXNXX,1,Macro(dialout-default,${EXTEN})
and the sip.conf looks like:
[200]
username=200
type=friend
secret=tryagain
qualify=no
port=5060
pickupgroup=
nat=never
mailbox=
host=dynamic
dtmfmode=rfc2833
disallow=
context=from-internal
canreinvite=no
callgroup=
callerid=Roaming SoftPhone 200
allow=
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RE: [Asterisk-Users] Netlogic inbound DID issue
Per Mike's issue here, we're noticing this problem with older versions
of Asterisk (it would seem?), and especially distrib [EMAIL PROTECTED] As
he stated we're seeing 'No Authority Found' coming from the clients, in
[EMAIL PROTECTED] we get see the No Authority found on the server, and the
client
sees absolutely nothing.
What's strange is I personally run CVS-head at my house, dated 11/10/04,
it has no problems at all.
If anyone has info on this please help, it's killing us :D
Matt
-Original Message-
From: Mike Clark [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 17, 2005 11:02 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Netlogic inbound DID issue
Anyone out there using NetLogic DIDs? And have inbound working? I got
outbound working, but no joy so far with inbound. Here are the relevant
parts from my conf files:
iax.conf
[general]
tos=lowdelay
jitterbuffer=no
register = username:[EMAIL PROTECTED]
[netlogic]
type=friend
host=dynamic
context=sourcekit-main
auth=plaintext
username=
secret=
disallow=all
allow=ulaw
allow=all
extensions.conf
[sourcekit-sip]
exten = 101,1,Dial(SIP/SK-101,20)
exten = 101,2,Voicemail(u101)
exten = 101,102,Voicemail(b101)
exten = 101,103,Hangup
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
[sourcekit-main]
include=sourcekit-sip
exten = +19193233010,1,GoTo(sourcekit-sip,101,1)
exten = _1NXXNXX,1,SetCallerID(9193233010)
exten = _1NXXNXX,2,Dial(IAX2/netlogic/${EXTEN})
exten =
_1NXXNXX,3,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,4,Hangup
[netlogic]
include=sourcekit-main
and, thr debug output from * CLI:
Asterisk Ready.
*CLI iax2 debug
IAX2 Debugging Enabled
*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX
Subclass: NEW
Timestamp: 00017ms SCall: 00030 DCall: 0 [206.80.70.49:4569]
VERSION : 2
CALLED NUMBER : +19193233010
Unknown IE 045 : Present
CALLING NUMBER : +13362150564
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
LANGUAGE: en
CALLED CONTEXT : netlogic
USERNAME: username
FORMAT : 4
CAPABILITY : 2097151
ADSICPE : 2
DATE TIME : 175199382
Ignoring unknown information element 'Unknown IE' (45) of length 1 Mar
17 12:35:19 NOTICE[21100]: chan_iax2.c:5419 socket_read: Rejected
connect at
tempt from 206.80.70.49, who was trying to reach '[EMAIL PROTECTED]'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Timestamp: 00018ms SCall: 2 DCall: 00030 [206.80.70.49:4569]
CAUSE : No authority found
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Re: [Asterisk-Users] Group Ring after Timeout
On Fri, 18 Mar 2005 13:57:37 +0100, Reuben Grech [EMAIL PROTECTED] wrote: Dear All, Would like to know what I should do to:: pickup call immediately and simultaneously Ring a Group, so that caller is listening to message whilst group phones are ringing and first one to pickup gets the call. The dial command will call more than one device eg: exten = 1234,1,Dial(SIP/1234SIP/2345) extension 1234 will now ring sip device 1234 and 2345 Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager API - Redirect command
I read the Wiki pages about the Redirect command, but,if I want to do a redirect into a MeetMe room, from a *remote* machine, how do I *query* Asterisk and get the Channel details?i.e the values for the Channel and ExtraChannel.I am using *SIP only*.Also, when redirected, one end Hangs up. Is this the intendedbehavior? Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom vs. Cisco IP Phones
2. Doesn't handle multiple lines nicely, will not jump to the next line, even if the same SIP registration is used, and you can't disable Call waiting. You can most certainly turn off call waiting. That's handled in Asterisk. We use AMP, and our configs allow us *70 to turn CW on, and *71 to turn it off. 3. They don't realy support their phones, unless there is a hardware problem. They don't support them with Asterisk, but if you don't tell them about it, they tend to be very good at working to resolve issues. We have a fleet of IP600s that are doing well. The configs take a little bit to wrap your head around, and there are just so many features that you can enable or disable that it's a lot of work to tweak them to exactly what you may like. We're considering having our in-house programmer write some sort of PHP app for a web-based config .xml generator. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI-like calls in the [globals] section
I'd like to set up some global parameters once at startup using an external program. (eg like one would with AGI) How can I do that ? Thanks, Thomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Parking a call in manager interface
Is it possible to park a call through the manager interface? If yes; how? Regards Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] About the weather..
There is a script on the [EMAIL PROTECTED] sourceforge list that reads the weather for you. Basically ftp's a text file from the BOM and then uses festival to read it out to you Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Friday, March 18, 2005 2:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] About the weather.. At 12:08 AM -0500 on 3/18/05, Kris Edwards wrote: Ok, I've been away from the list for sometime now and feel as though I'm going to ask something that's been asked many times before, yet I find nothing in the wiki, so here goes: I notice that allison has done recordings for weather forecasts, yet I find no agi's that parse forecasts and use this recordings to piece together the forecast. I find plenty of weather agi's, but they all seem to be festival related rather than using prerecorded bits and pieces. Anybody know of a script that makes use of these recordings? Thanks for the help! To my knowledge, they were recorded in anticipation of somebody getting ambitious and writing some AGIs around the recordings. Perhaps you can try your hand at some Perl... JT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P Not loading Drivers
Thankyou. That did the trick. I must have got confused with zaptel and zapata conf files with an example. Regards, Greg On 18/03/2005, at 10:38 PM, Rich Adamson wrote: Inline... I am trying to get the drivers working with this device with 4 fxo modules on it. I do a modprobe zaptel and no errors appear. But when I do modprobe wctdm the following errors appear: Notice: Configuration file is /etc/zaptel.conf line 4: Cannot get number of tones chanel 1 line 4: Cannot init tones chanel 1 line 4: Cannot get number of tones chanel 2 line 4: Cannot init tones chanel 2 line 4: Cannot get number of tones chanel 3 line 4: Cannot init tones chanel 3 line 4: Cannot get number of tones chanel 4 line 4: Cannot init tones chanel 4 8 error(s) detected /lib/modules/2.4.22-2f/misc/wctdm.o: post-install wctdm failed /lib/modules/2.4.22-2f/misc/wctdm.o: insmod wctdm failed I have googled the mailing lists and have not been able to find anything on these errors. My config of zaptel.conf fxsks=1-4 loadzone=au defaultzone=au channels=1-4 Unless the above is a typo, channels=1-4 should not be in this file; only the first three lines zapata.conf [channels] language=en ; XTDM20B Port #1,2 plugged into PSTN ; context=from-1800 signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no group=1 channel=1 Not sure about the syntax of channel=1; I know channel=1 works though. context=from-pstn signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=3 channel=2-3 context=from-fax signalling=fxs_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=3 channel=4 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail.conf extractor?
Hi, I believe there is a script that reads the contents of voicemail.conf and goes on to send the voice e-mail messages to whatever e-mail address specified in voicemail.conf. What's the name of this script and where is it located? Thanks, Julius. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which linux distribution
Hi all i'm just starting to setup my "own" asterisk. My first question is, if there is any reason to choose aspecial linux distribution or if it doesn't mater which distribution i chosse. Is there anything i should be aware of? Thanks a lot for your help! Greetings Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Problem = Segmentation faults Seems to be asterisk
Jose R. Ortiz Ubarri wrote: Hi: I had asterisk with RealTime database working perfectly in a RH 9.0 machine. I used the sip cache so I even had MWI working. The problem is that I decided to move to Fedora Core 3. I installed the lastets cvs version of asterisk and the RealTime addon from asterisk-addons. I at first had the problems with the kernel and the zaptel driver but all that was solved with the configuration from the Asterisk Wiki. Then when I moved my configuration to the new asterisk server and configured the RealTime addon it falls in a Segmentation fault. If I do not load the res_config_mysql.so (edited at modules.conf) then asterisks runs without any problem. But if I load the module from boot or from the asterisk command load res_config_mysql.so then I get the Segmentation fault again. I'm not sure what the problem is. Is it a Fedora Core 3 problem, or an Asterisk latest version problem? I don't think it is a configuration problem because I just used the same configuration I had before. The only diferences may be the OS and probably the asterisk version that is only one week newer than the one I was running in the old asterisk server, so I'm probably even running the same version of asterisk in both machines. Any advise? Someone else have a similar configuration working with Fedora Core 3? Thanks in advance, Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? -- Jose R. Ortiz Ubarri (CHEO), CS System Administrator / Programmer High Performance Computing facility - UPR Email: [EMAIL PROTECTED]|[EMAIL PROTECTED] Phone: 787-758-3054 Fax: 787-758-3058 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P install problems
Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P install problems
If you have any FXS ports, use wcfxs. On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which linux distribution
If it's your very first * setup, use [EMAIL PROTECTED] -- comes on an ISO with its own distro of CentOS and should make initial setup quite easy. Other than that, search the list archives (google with site:digium.com) for your favorite distro and see what issues you uncover. Right up front -- if you don't have digium hardware nor a proper USB port (search for ztdummy), you may want a 2.6 kernel such as found in FC3. -Original Message- From: Frank Fischer [mailto:[EMAIL PROTECTED] Sent: Friday, March 18, 2005 7:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Which linux distribution Hi all i'm just starting to setup my own asterisk. My first question is, if there is any reason to choose a special linux distribution or if it doesn't mater which distribution i chosse. Is there anything i should be aware of? Thanks a lot for your help! Greetings Frank ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] TDM400P install problems
Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme2 compilation problem
Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:317: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:317: for each function it appears in.) make: *** [app_dial.o] Error 1 Thanks in advance Anil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX Registration being lost
I would assume this is an issue of port forwarding too. Without having a statically mapped port forward from your firewalls external to the * internal IP, the port can shift. The answer I gave below should resolve your problem as well. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Thursday, March 17, 2005 4:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX Registration being lost Speaking of this. Why is it that sometimes the port is shown as something differente than 4569 on some hosts? For ex. Host UsernamePerceived Refresh State 210.80.176.12:221108990608214 1.2.3.4:4569 60 Registered And why that host changes port each time it reboots? This is happening to one IAX box I connect to.. And it's a pain cause I have to put the port on my dial and registry entries in order to register on it or dial to it. Why is that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Jueves, 17 de Marzo de 2005 03:08 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAX Registration being lost Tony, Do you have port 4569 on your external firewall IP port-forwarded to your internal IP on the * box? You should create a port forward of the external eth1:4569 -- 192.168.100.183:4569 Assuming that you exxternal IP were something like 1.2.3.4, you should see this when you run iax2 show registry. Host UsernamePerceived Refresh State 210.80.176.12:45698990608214 1.2.3.4:4569 60 Registered Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Davidson Sent: Thursday, March 17, 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX Registration being lost Well, this is getting more interesting. I started looking at this this morning and realised that Asterisk had lost registration, yet my ADSL connection has been up for almost 2 days - and it was working fine yesterday. Therefore it doesn't appear to be related to the IP address changing. I'm thinking it's more that the registration is lost for any reason (such as an ADSL reconnect or the registration needing to be refreshed) and it won't come back. Get this message as before: Host UsernamePerceived Refresh State 210.80.176.12:45698990608214 Unregistered 60 Request Sent I tried a ping and a traceroute and both working fine. An ifconfig just shows the internal address (192.168.100.183). Tony Davidson CNA CA (IT) DCE Director, Zero Effort Networking Pty Ltd Ph: 0411 478 004, Fax: (02) 8569 2012 http://www.zeroeffortnetworking.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, 17 March 2005 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAX Registration being lost What doesn't make sense about that is that if you are setup like this... DSL Router --- Your Firewall/Router --- Asterisk Box Then the issue of being dynamic will not matter to the * box. IP storing is mute since the end point and start point are not changing. All that is changing is the IP on the outside of your Firewall/Router and thus a momentary loss of connectivity. AAH would not care about that in relation to what it has stored. It will just attempt the registration and pass data to the gateway (inside interface of your FW/Router) just like before. As far as it is concerned, nothing has changed except now the attempt to communicate outward dies on the first hop until the new IP is assigned to the external interface of your FW. Try this. Start some IAX debug in the CLI the next time it happens. Tracert your IAX target and see if you can get to it. Ifconfig the interface to see what is setup. Report back. Thanks, Wiley My mailbox is spam-free with ChoiceMail, the leader in personal and corporate anti-spam solutions. Download your free copy of ChoiceMail from www.choicemailfree.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
Re: [Asterisk-Users] Extension ringing but no ringing sound asterisk
Paul Dracevich wrote: When I call from extension A on Box and to Extension A on Box B I get no ringing sound. That is under your control. The Dial command has options and one of the options (r) is to add ringing sound. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Manager API - Redirect command
You should be able to get the full channel values by doing a "Action: Command Command: Show Channels" and picking your SIP extension out of the list it gives you of active channels. Then you can take that and the channel that you are currently connected to, also taken from the "Show channels" output(I am assuming that you want to take both parties and dump them into the meetme room) and use those two channel values in the Redirect command. If you do it right, neither you nor the person you were talking to would notice that you just moved into the meetme room. We use this method with the astGUIclient client application to transfer an existing conversation into a meetme room and it works great. Hope this helps, MATT--- -Original Message-From: Vyom A [mailto:[EMAIL PROTECTED]Sent: Friday, March 18, 2005 8:21 AMTo: Asterisk_users_mailing_listSubject: [Asterisk-Users] Manager API - Redirect command I read the Wiki pages about the Redirect command, but,if I want to do a redirect into a MeetMe room, from a *remote* machine, how do I *query* Asterisk and get the Channel details?i.e the values for the Channel and ExtraChannel.I am using *SIP only*.Also, when redirected, one end Hangs up. Is this the intendedbehavior? Do you Yahoo!?Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BV This morning
Last night everything was working perfectly. This morning I went to make a call and got a message "the device you are using is not registered to make calls on the network". Connecting direct to BV on X-Lite works fine. Any ideas? -Kerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] TDM400P install problems
Can you run dmesg after that command and tell us what the relevant output is? On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie can't dial out to pstn
What version of Asterisk? If this is not [EMAIL PROTECTED] you may want to
install it and start over. It eases many of the problems experienced by newbs
when learning *.
Otherwise, make sure you use the ztcfg - so you can see some error
verbosity.
You may need to recompile your zaptel stuff. Just make sure you follow the
instructions and recompile asterisk after.
Regards,
Wliey
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg
Sent: Thursday, March 17, 2005 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Shane Dalgleish
Subject: Re: [Asterisk-Users] Newbie can't dial out to pstn
I have just run ztcfg and got these errors:
# ztcfg
Notice: Configuration file is /etc/zaptel.conf line 209: Cannot get number of
tones chanel 1 line 209: Cannot init tones chanel 1 line 209: Cannot get number
of tones chanel 2 line 209: Cannot init tones chanel 2 line 209: Cannot get
number of tones chanel 3 line 209: Cannot init tones chanel 3 line 209: Cannot
get number of tones chanel 4 line 209: Cannot init tones chanel 4
What would these mean. I searched the archives and couldn't find these errors.
Greg
On 18/03/2005, at 1:24 PM, Greg wrote:
I was just copy an example from somewhere. I made the change but the
mobile still doesn't ring. The line the card is attached to works
fine. here is the new output
Executing Goto(SIP/2002-4385, mobile|0400039953|1) in new stack
-- Goto (mobile,0400039953,1)
-- Executing Goto(SIP/2002-4385, localcall|0400039953|1) in
new stack
-- Goto (localcall,0400039953,1)
-- Executing Dial(SIP/2002-4385, ZAP/1/0400039953|60|r) in new
stack
-- Called 1/0400039953
-- Zap/1-1 answered SIP/2002-4385
-- Hungup 'Zap/1-1'
== Spawn extension (localcall, 0400039953, 1) exited non-zero on
'SIP/2002-4385'
is this line -- Zap/1-1 answered SIP/2002-4385 displayed when the card
tries to make the call or when the card thinks it has established the
call?
Regards,
Greg
By the way, I'm on the Gold Coast.
On 18/03/2005, at 12:32 PM, Shane Dalgleish wrote:
Greg,
Any reason why you are putting the country code on the front for a
mobile call through pstn?
(Unless you have something like an Ericsson F220M Fixed Cellular
Terminal connected to it?)
And you said the tdm400p never tries to pick up the phone?
Have you connected a normal phone on the line and had a listen?
Where is Aus are you? :o)
Cheers
Shane
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Sent: Friday, 18 March 2005 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Newbie can't dial out to pstn
Hi,
I have just put in a tdm400p with 4 fxo modules and am trying to
dial out from x-lite to dial my mobile phone just to test.
The output in the asterisk console is like this
Executing Goto(SIP/2002-239b, mobile|61400039953|1) in new stack
-- Goto (mobile,61400039953,1)
-- Executing Goto(SIP/2002-239b,
localcall|61400039953|1) in new stack
-- Goto (localcall,61400039953,1)
-- Executing Dial(SIP/2002-239b,
ZAP/1/61400039953|60|r) in new stack
-- Called 1/61400039953
-- Zap/1-1 answered SIP/2002-239b
-- Hungup 'Zap/1-1'
== Spawn extension (localcall, 61400039953, 1) exited non-zero on
'SIP/2002-239b'
It never tries to pick up the phone and dial out. I'm not sure if
the config is correct, but I can easily dial between x-lite clients,
just not get the pstn.
Can anyone see any glaring mistakes?
Any help is grealty appreciated.
Regards,
Greg
My extensions.conf part is this:
exten = _04,1,GoTo(mobile,61${EXTEN:1},1)
[localcall] ; local calls by PSTN ?is a fixed charge, voip is per
minute exten = _X.,1,Dial(ZAP/1/${EXTEN},60,r) exten =
_X.,2,Congestion exten = _X.,3,Hangup exten = _X.,103,Hangup exten
= _X.,104,Hangup exten = _X.,105,Hangup
[mobile] ; Maybe be cheaper to route mobile calls differently to STD
in some cases exten = _X.,1,Goto(localcall,${EXTEN},1)
zaptel.conf
fxsks=1-4
loadzone=au
defaultzone=au
channels=1-4
zapata.conf
[channels]
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
pickupgroup=1-4
immediate=no
context=incomingcall
signalling=fxs_ks
callerid=asreceived
channel=1-4
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[Asterisk-Users] Meetme2 compilation Err
Hi , While compiling meetme2 i am getting the following error. Please guide me to make it work. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:317: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:317: for each function it appears in.) make: *** [app_dial.o] Error 1 Thanks in advance Anil ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm cannot be found in any file form... but it's there
Are you sure it is in the right directory. Perhaps mentioning the directory you placed the file in and listing the section of your dialplan would go a long in helping someone help you. Umar On Fri, 18 Mar 2005 06:11:36 +, Scheda [EMAIL PROTECTED] wrote: Hey, I recorded this intro, and changed it to a gsm file in the shell, and I'm getting an error saying that it isn't in the directory at all when it's sitting right there. I don't know why that is. If you want to hear it, it's http://scheda.underfireradio.com/astintro.mp3 I don't know what the matter is, I've tried renaming it, copy and pasting it in there, deleting it and placing it back... I'm kinda out of ideas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 convert to sip
Hi This is found on cisco.com. http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml hope it helps Regards, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Krasavin Andrey Sent: Friday, March 18, 2005 7:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7940 convert to sip Hi! Can anybody help me with convert Cisco 7940 CallManager Phone to a SIP Phone? I have continious error in tftp log: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10081]: peer requests OS79XX.TXT, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10081]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10080 from=192.168.1.111 Mar 18 15:12:30 AKrasavin xinetd[10068]: EXIT: tftp pid=10080 duration=0(sec) Mar 18 15:12:30 AKrasavin xinetd[10068]: START: tftp pid=10082 from=192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10083]: connect from 192.168.1.111 Mar 18 12:12:30 AKrasavin utftpd[10083]: peer requests SEP000AF4BB7D59.cnf.xml, conversion octet Mar 18 12:12:30 AKrasavin utftpd[10083]: unterminated option value in init packet Mar 18 15:12:30 AKrasavin xinetd[10068]: EXIT: tftp pid=10082 duration=0(sec) OS79XX.TXT and SEP000AF4BB7D59.cnf.xml exists. I'll be very thankful for any your help. -- WBR, Krasavin Andrey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] limit about asterisk pstn out
Set this up under your own extension where the first line shall read as
exten = [EMAIL PROTECTED]/_9NXXNX,1,Congestion
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FCG ZHAO
Zigang
Sent: Friday, March 18, 2005 1:17 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] limit about asterisk pstn out
I have a system include asterisk + ser.
when I want to limit a dial out to pstn , I will do that :
extensions.conf
exten = _9NXXNX/[EMAIL PROTECTED],Congestion
exten = _9NXXNX, 1,Dial(ZAP/g2/{EXTEN:1},30,t) exten =
_9NXXNX, 2,Hungup
but I don't confirm is it right.
I have no env to test it.
who can help me?
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Re: [Asterisk-Users] gsm cannot be found in any file form... but it's there
Scheda wrote: Hey, I recorded this intro, and changed it to a gsm file in the shell, and I'm getting an error saying that it isn't in the directory at all when it's sitting right there. I don't know why that is. Check that the file permissions are correct (owner/group/permissions match other files). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX Registration being lost
I agree with you Wiley. Just wanted to make sure I was on the right path. Thx -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Viernes, 18 de Marzo de 2005 08:38 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAX Registration being lost I would assume this is an issue of port forwarding too. Without having a statically mapped port forward from your firewalls external to the * internal IP, the port can shift. The answer I gave below should resolve your problem as well. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Thursday, March 17, 2005 4:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX Registration being lost Speaking of this. Why is it that sometimes the port is shown as something differente than 4569 on some hosts? For ex. Host UsernamePerceived Refresh State 210.80.176.12:221108990608214 1.2.3.4:4569 60 Registered And why that host changes port each time it reboots? This is happening to one IAX box I connect to.. And it's a pain cause I have to put the port on my dial and registry entries in order to register on it or dial to it. Why is that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Jueves, 17 de Marzo de 2005 03:08 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAX Registration being lost Tony, Do you have port 4569 on your external firewall IP port-forwarded to your internal IP on the * box? You should create a port forward of the external eth1:4569 -- 192.168.100.183:4569 Assuming that you exxternal IP were something like 1.2.3.4, you should see this when you run iax2 show registry. Host UsernamePerceived Refresh State 210.80.176.12:45698990608214 1.2.3.4:4569 60 Registered Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Davidson Sent: Thursday, March 17, 2005 1:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX Registration being lost Well, this is getting more interesting. I started looking at this this morning and realised that Asterisk had lost registration, yet my ADSL connection has been up for almost 2 days - and it was working fine yesterday. Therefore it doesn't appear to be related to the IP address changing. I'm thinking it's more that the registration is lost for any reason (such as an ADSL reconnect or the registration needing to be refreshed) and it won't come back. Get this message as before: Host UsernamePerceived Refresh State 210.80.176.12:45698990608214 Unregistered 60 Request Sent I tried a ping and a traceroute and both working fine. An ifconfig just shows the internal address (192.168.100.183). Tony Davidson CNA CA (IT) DCE Director, Zero Effort Networking Pty Ltd Ph: 0411 478 004, Fax: (02) 8569 2012 http://www.zeroeffortnetworking.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, 17 March 2005 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IAX Registration being lost What doesn't make sense about that is that if you are setup like this... DSL Router --- Your Firewall/Router --- Asterisk Box Then the issue of being dynamic will not matter to the * box. IP storing is mute since the end point and start point are not changing. All that is changing is the IP on the outside of your Firewall/Router and thus a momentary loss of connectivity. AAH would not care about that in relation to what it has stored. It will just attempt the registration and pass data to the gateway (inside interface of your FW/Router) just like before. As far as it is concerned, nothing has changed except now the attempt to communicate outward dies on the first hop until the new IP is assigned to the external interface of your FW. Try this. Start some IAX debug in the CLI the next time it happens. Tracert your IAX target and see if you can get to it. Ifconfig the interface to see what is setup. Report back. Thanks, Wiley My mailbox is spam-free with ChoiceMail, the leader in personal and corporate anti-spam solutions. Download your free copy of ChoiceMail from www.choicemailfree.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re[4]: [Asterisk-Users] TDM400P install problems
Hello Dana, Friday, March 18, 2005, 3:40:21 PM, you wrote: DO Can you run dmesg after that command and tell us what the relevant output is? # modprobe zaptel modprobe wcfxs FATAL: Could not open '/lib/modules/2.6.9-1.667/misc/wctdm.ko': No such file or directory # dmesg Zapata Telephony Interface Registered on major 196 # I have to say that there are 2 cards in this server, this is my zaptel.conf fxoks=32-35 loadzone = us defaultzone = us span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 was running cvs-head, now running 1.0.6 It seems that when I call wcfxs wctdm is called instead. Any idea ? TNX ! DO On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to place calling rule contexts?
Matt wrote: If I only want to give my sip users say local calling where do I put that in the sip config? ... and the sip.conf looks like: [200] ... context=from-internal ... The context=from-internal is the key. You will need to create a context called from-internal that only includes local calling. For example: [from-internal] include = outbound-local include = internal-extensions include = outbound-emergency ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redhat 9 Music on hold
Jason Becker wrote: Daniel Burget wrote: I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere, nothing seems to help. I am using 59r of MPG123, and do not have MPG321 installed. I did a 'make mpg123' from asterisk, make no difference. I believe it is a bug: http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/85000 although I don't know if a bug was ever filed. I had a cursory look at the time we were bitten by this but couldn't find one. Pulling a newer CVS Stable and rebuilding resolved the issue. And if you are on the asterisk-cvs mailing list, you would have seen a fix being added yesterday. See: http://www.lists.digium.com/ --Eric -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme2 compilation problem
I'm sure there was a patch for meetme2 regarding compilation... google for meetme2 + patch. It worked for me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Undocumented exten syntax?
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten = s,1,SetVar(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,SetGlobalVar(EMERGENCY=1)
exten = s,n,SetVar(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY} = 1]?inprogress)
I hope the wiki page mentions that the n priority is only supported
in CVS-HEAD, not 1.0.x.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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[Asterisk-Users] newbe question sip.conf
Dear Gurus, Just installed Asterisk and it runs just fine. Have made a simple extension and sip configuration which works nice also. But still a question. My (simple) extension is as follows: extensions.conf: [default] extern = 1001,1,Dial(SIP/1001) extern = 1002,1,Dial(SIP/1002) and the sip.conf: [1001] type=friend host=dynamic canreinvite=no disallow=all allow=alaw [1002] ; copy of 1001 type=friend host=dynamic canreinvite=no disallow=all allow=alaw So far, so good. But if I would like to deploy e.g. 100 sip phones, I would have to add 100 sections? What I would like to do, is group 'm like: extensions.conf: [default] exten = _1XXX,1,Dial(SIP/norm,20) and in the sip.conf: [norm] type=friend host=dynamic canreinvite=no disallow=all allow=alaw But this doesn't seem to work. Any suggestions? Thanks Martin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom vs. Cisco IP Phones
If you've considered the Snom, you might also want to test a Zultys 4x4 or 4x5. I picked a 4x5 up off of ebay recently and have been pleasantly surprised by it. While I don't currently have a Polycom to compare it with, I would rank the audio quality equal to the Cisco's. It also just 'does the right thing' with multiple lines - only one registration, no hints needed. Can be configured through TFTP with both default and phone specific config files. Software updates are freely available from the Zultys website. I took a look at the Zultys phones when I was first shopping around. One of their reps was kind enough to lug an entire working phone setup into our office. He had some 4x4's and 4x5's and also a Cisco 7960 (just to show that their system was open standards compliant). I liked the 4x5's ease of use, the 4 port network switch, the native PoE, and the hard buttons for holding and transferring. Much to his chagrin, though, I was actually much more impressed by the 7960. The 4x4's and 4x5's just looked like lower quality equipment. I suppose it didn't help that the plastic casing on his 4x4 was cracked and broken. In the end I went with neither, though, because the Polycom units were so much cheaper. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] leaky reload
Thomas Andrews wrote: If I comment out the following line in zapata.conf I would expect asterisk to forget the cli information for that channel when I reload: callerid=Uniden Dead (256) 428-6125 ... but it doesn't; I have to restart asterisk for it to take effect. The funny thing is that the reverse is *not* true - ie if I uncomment the line and reload then it learns about the caller id Uniden Dead. Why is this a one-way process ? Issuing a reload to asterisk does not correctly reload /etc/asterisk/zapata.conf. This has been the case forever. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbe question sip.conf
Hi, -Original Message- Just installed Asterisk and it runs just fine. Have made a simple extension and sip configuration which works nice also. But still a question. My (simple) extension is as follows: extensions.conf: [default] extern = 1001,1,Dial(SIP/1001) extern = 1002,1,Dial(SIP/1002) and the sip.conf: [1001] type=friend host=dynamic canreinvite=no disallow=all allow=alaw [1002] ; copy of 1001 type=friend host=dynamic canreinvite=no disallow=all allow=alaw So far, so good. But if I would like to deploy e.g. 100 sip phones, I would have to add 100 sections? Yup, you would. Which is why we all download or develop tools to automate that kind of thing :) Best regards, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom vs. Cisco IP Phones
On Mar 18, 2005, at 8:27 AM, Ben Ruset wrote: 3. They don't realy support their phones, unless there is a hardware problem. They don't support them with Asterisk, but if you don't tell them about it, they tend to be very good at working to resolve issues. You have to know what issues they consider to be related to the platform. In general, copper and plastic issues (i.e., the phone is in the wrong number of pieces) the direct customer support people can help you with. As soon as you start talking about configuration, though, don't bother trying to get any specifics out of them that aren't in the admin guide -- you'd be wasting your time (as have many on this list before you). -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme2 compilation problem
I did the patch also . That didnt help me. I am using CVS head of 17th March . Googling didnt give me much info other than this patch. Thanks On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell [EMAIL PROTECTED] wrote: I'm sure there was a patch for meetme2 regarding compilation... google for meetme2 + patch. It worked for me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Group Ring after Timeout
Dear All, I am listening to blips during conversations when I have an incoming call from an X100P card. This does not happen on all conversations. Any clues? :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme2 compilation problem
Just use [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ Meetme2 is automatically installed Cheers dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anil Kumar K Sent: Friday, March 18, 2005 10:56 AM To: Giovanni Powell Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme2 compilation problem I did the patch also . That didnt help me. I am using CVS head of 17th March . Googling didnt give me much info other than this patch. Thanks On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell [EMAIL PROTECTED] wrote: I'm sure there was a patch for meetme2 regarding compilation... google for meetme2 + patch. It worked for me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to place calling rule contexts?
Got it.. thanks that worked... On Fri, 18 Mar 2005 09:12:34 -0600, Scott Nelson [EMAIL PROTECTED] wrote: Matt wrote: If I only want to give my sip users say local calling where do I put that in the sip config? ... and the sip.conf looks like: [200] ... context=from-internal ... The context=from-internal is the key. You will need to create a context called from-internal that only includes local calling. For example: [from-internal] include = outbound-local include = internal-extensions include = outbound-emergency ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk reload
I have read gracefully restarting asterisk on a regular basis is a good idea. However the problem I have with doing this is that I need to have all my users log back in , using AgentCallbackLogin and AddQueueMember. Is there any way anyone has come up with to keep the state of all users between restarts. I should probably also mention that all the agents have there own passwords. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[4]: [Asterisk-Users] TDM400P install problems
Try using module wctdm instead. That solved a lot of headaches for me. On Fri, 18 Mar 2005 16:02:23 +0100, Alessio Focardi [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:40:21 PM, you wrote: DO Can you run dmesg after that command and tell us what the relevant output is? # modprobe zaptel modprobe wcfxs FATAL: Could not open '/lib/modules/2.6.9-1.667/misc/wctdm.ko': No such file or directory # dmesg Zapata Telephony Interface Registered on major 196 # I have to say that there are 2 cards in this server, this is my zaptel.conf fxoks=32-35 loadzone = us defaultzone = us span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 was running cvs-head, now running 1.0.6 It seems that when I call wcfxs wctdm is called instead. Any idea ? TNX ! DO On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Scott Griepentrog ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAC Access Bank Manual
On Mar 18, 2005, at 2:40 AM, George Pajari wrote: Vicky Shrestha wrote: The asterisk configuration and the channel bank configuration are both set to esf and b8zs. Howerver I am still getting the framing Error Red and blinking. zttool shows there are no alarms. According to the manual, Framing Error (Red and Blinking )means Network T1 is out of frame (received signal cannot be framed to ESF or D5 as configured by T1 Option switch 4) I tried with both DIP switch on and off, but no help. Any ideas ? Is my card or channel bank bad ? Probably not. More likely your CAC ABI is set up to run in TR08 mode which is incompatible with standard T1 framing. Check the LIU board (the Line Interface Board -- as opposed to the FXO or FXS cards) and look at the PROM. It usually will be marked TR08. If that is the case you will need to order a D4/ESF upgrade kit. If you have a 1.x revision TR08 chip, you will need P/N 750-0018. If you have a 3.x revision TR08 chip, you will need P/N 750-0019. Are you using a terminal to talk to the CAC? Depending on configuration you may have the DIP switches disabled and changing them will do no good. Connect to the CAC and ask it for the T1 configuration to verify what it really is. Also make sure you rerun ztcfg after any changes to zaptel.conf. What does zttool tell you? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reply a post
Hi how do i reply a question asked in this mailling list. tks Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk reload
CVS head has an option to do this. persistentmembers is the option I think. Julian LS. James Murray wrote: I have read gracefully restarting asterisk on a regular basis is a good idea. However the problem I have with doing this is that I need to have all my users log back in , using AgentCallbackLogin and AddQueueMember. Is there any way anyone has come up with to keep the state of all users between restarts. I should probably also mention that all the agents have there own passwords. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reply a post
Do you know how to hit the reply button on the Outlook menu? Just hit the reply button. If you dont know this, send an email to asterisk-users@lists.digium.comif it is a user related topic. Dont post business topics here. Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanishka SomaratneSent: Friday, March 18, 2005 11:15 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] reply a post Hi how do i reply a question asked in this mailling list. tks Kanishka NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk reload
Asterisk wrote: CVS head has an option to do this. persistentmembers is the option I think. Yes, CVS HEAD has both persistent dynamic members and persistent agent logins. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Meetme2 compilation problem
Giovanni,on ftp://ftp.vinkconsult.com/downloadsis a patched version of app_meetme2.c.I patched and compiled it against the CVS unstable from todayAndre- Oorspronkelijk Bericht -Onderwerp:Re: [Asterisk-Users] Meetme2 compilation problemAfzender: Anil Kumar K [EMAIL PROTECTED]Aan:Giovanni Powell [EMAIL PROTECTED]CC:asterisk-users@lists.digium.comDatum:18-03-2005 16:56I did the patch also . That didnt help me. I am using CVS head of 17th March .Googling didnt give me much info other than this patch.Thanks< br />On Fri, 18 Mar 2005 10:18:26 -0500, Giovanni Powell[EMAIL PROTECTED] wrote: I'm sure there was a patch for meetme2 regarding compilation... google for meetme2 + patch. It worked for me.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo / delay problem
I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was rxgain and txgain. below is my current zapata.conf file All help would be grateful. I have tried and tried for 2 weeks it is rather annoying and irating to hear this delay/echo I would call it a delay since you can hear the end of the sentence repeat over and over. Also every now and again it sounds like a underwater submarine with ping and all. Thanks in advance Barry [channels] language = en context = inbound signalling= fxs_ks usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres= yes callwaitingcallerid = yes threewaycalling = yes echocancel= 16 echocancelwhenbridged = yes echotraining = no ;; yes rxgain= -2.0 txgain= -2.0 musiconhold = default channel = 1 context = intern signalling= fxo_ks callwaiting = yes usecallerid = yes echotraining = no ;; yes echocancel= 16 channel = 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Group Ring after Timeout
On Fri, 2005-03-18 at 16:59 +0100, Reuben Grech wrote: Dear All, I am listening to blips during conversations when I have an incoming call from an X100P card. This does not happen on all conversations. Any clues? :) Turn off call waiting in zapata.conf callwaiting=no -Seth Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call a url and get a result in the dialplan
Hi, can a call a php script wich is located in a remote server, someting like calling www.theserveraddress.com/scripts/validate?code=234234swq and get the result which this script generates (a 0 or a 1) back in the dial plan in a direct way or should I create a script which in turn does this? I'm using * CVS HEAD. Also I searched for this for I while but didn't manage to find anything but SendUrl and PHP AGI thanks M. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[6]: [Asterisk-Users] TDM400P install problems
Hello Scott, Friday, March 18, 2005, 5:10:14 PM, you wrote: SG Try using module wctdm instead. That solved a lot of headaches for me. There is no wctdm module in zaptel-1.0.6.tar.gz . So why when I call wcfxs ... modprobe wcfxs FATAL: Could not open '/lib/modules/2.6.10-1.770_FC3/misc/wctdm.ko': No such file or directory That does not look normal to me, I have built another kernel to try to make this behavior go away, still no luck Tnx anyway ... SG On Fri, 18 Mar 2005 16:02:23 +0100, Alessio Focardi SG [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:40:21 PM, you wrote: DO Can you run dmesg after that command and tell us what the relevant output is? # modprobe zaptel modprobe wcfxs FATAL: Could not open '/lib/modules/2.6.9-1.667/misc/wctdm.ko': No such file or directory # dmesg Zapata Telephony Interface Registered on major 196 # I have to say that there are 2 cards in this server, this is my zaptel.conf fxoks=32-35 loadzone = us defaultzone = us span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 was running cvs-head, now running 1.0.6 It seems that when I call wcfxs wctdm is called instead. Any idea ? TNX ! DO On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hello Dana, Friday, March 18, 2005, 3:23:36 PM, you wrote: DO If you have any FXS ports, use wcfxs. No, only green modules. But this is what I get when loading driver modprobe wcfxs FATAL: Error inserting wctdm (/lib/modules/2.6.9-1.667/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm What relates wcfxs to the wctdm that I was using previously ? Maybe deleting wctdm DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi DO [EMAIL PROTECTED] wrote: Hi, I was using a TDM400P with cvs version of asterisk, loading the driver with modprobe wctdm. Some days ago I switched to stable version 1.0.6, where I found no trace of such module ... is wcfxo to be used instead ? Do I also have to change something in zaptel.conf ? Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] DO ___ DO Asterisk-Users mailing list DO Asterisk-Users@lists.digium.com DO http://lists.digium.com/mailman/listinfo/asterisk-users DO To UNSUBSCRIBE or update options visit: DOhttp://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I4l + HiSax
I need HELP pls! BRISTUFF: Bad Sound quality CAPI: PTP Mode dont supported mISDN : kernel is 2.4.x and not 2.6.x HISAX : PTMP ok, PTP incoming ok but in outgoing asterisk dont compose number(i listen dial tone and than i can compose number via dtmf) Asterisk CLI (g3 is group of Modem[i4l]/ttyI0 and ttyI1): Called g3: 345344 (Channel is used but dont compose number) Asterisk Log: VERBOSE[10406]: -- Called g3:345344 Mar 18 23:41:01 DEBUG[10406]: Detecting DTMF inband with sw DSP on /dev/ttyI1 Mar 18 23:41:01 DEBUG[10406]: Dropping duplicate answer! Mar 18 23:41:01 VERBOSE[10406]: -- Modem[i4l]/ttyI1 answered SIP/201-3e46 Mar 18 23:41:01 DEBUG[10406]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 55994: Found Mar 18 23:41:01 DEBUG[10406]: Ooh, format changed from unknown to alaw Mar 18 23:41:06 DEBUG[10406]: Didn't get a frame from channel: SIP/201-3e46 Mar 18 23:41:06 DEBUG[10406]: Bridge stops bridging channels SIP/201-3e46 and Modem[i4l]/ttyI1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI Cause Code Help
Peter Svensson wrote: The two issues are only somewhat related. The RELEASE COMPLETE as an reply to a SETUP after having sent a CALL PROCEEDING is probably not allowed by the state transitions listen in q.931. I've commented out a few lines of code to make sure * sends DISCONNECT but I'm getting identical results. Seems like it doesn't matter if I skip to RELEASE_COMPLETE or not. The in-band announcement is more related to whether we have sent a progress information element which states that in-band audio is available. I think Asterisk sends such a progress message almost as soon as possible. However, in this case the problem is a CALL PROCEEDING before the RELEASE_COMPLETE answering teh SETUP. The fact that the CALL PROCEEDING also includes a PROGRESS element is incidental. Are you suggesting that * is telling the other side that we are making noise and they shouldn't? My intent here is to have the telco say I'm sorry the numbern is not in service instead of tying up one of our lines for the duration of such a message. Likewise with Congestion odd part is that Busy works fine, as I mentioned in my original post. Thanks for the posts Peter and Eric. Good to know I am going about this in the right direction. Trevor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Group Ring after Timeout
Many thanks will try that out! :) Could a similar setting also cause telephone lines to drop or the callers to hear me very far away?? Thanks Again -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: 18 March 2005 17:23 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Group Ring after Timeout On Fri, 2005-03-18 at 16:59 +0100, Reuben Grech wrote: Dear All, I am listening to blips during conversations when I have an incoming call from an X100P card. This does not happen on all conversations. Any clues? :) Turn off call waiting in zapata.conf callwaiting=no -Seth Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 190 Loud Ring While on Speaker
When on a speaker call on the SNOM 190, a second calls comes in and the ring is VERY loud and heard by the remote party. Is there a way to set the dialplan in * for a silent ring for the second (or third) call. Is there a way on the snom to change the ring to a silent ring when on speaker? Has anyone else had this experience? TIA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error message
When issuing a "stop now" to Asterisk, the message "Yuck! Error in buffer handling...: Success" is returned. No complaints when Asterisk is started, and everything seems OK while running . Google provides no help RH9, CVS-HEAD of 2/24/2005 Any clue as to why? John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users]ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules)
Thanks for all of your help Dan, I will continue to try to figure out why this will not patch. I installed it on a clean * box and it works fine, It just won't on the boxes with [EMAIL PROTECTED] When I find the answer I will post so if others run into this problem there will be a solution. Thanks again Henry - Original Message - From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 18, 2005 3:03 AM Subject: RE: [Asterisk-Users]ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules) I'm afraid I am at a loss. If the three files, app_cbmysql.c, app_meetme2.c and Makefile all exist in ../apps then a patch -p1 from the ../asterisk directory should work. The -p1 tells patch to ignore the first directory in the path to the file in the patch, -p2 ignores two directories. Another option is to just edit the apps-meetme-cbmysql.txt and split it into three patchs and apply them one at a time. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Thursday, March 17, 2005 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ANNOUNCEMENT:Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules) It doesn't matter if I run it from the apps directory or the asterisk directory I get the same response. This is getting frustrating. - Original Message - From: Dan Austin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 18, 2005 12:17 AM Subject: RE: [Asterisk-Users] ANNOUNCEMENT: Updatesforapp_cbmysqlandMeetMe2gui(out of tree modules) I should have read a little closer- [EMAIL PROTECTED] apps]# patch -p1 /var/build_aah/asterisk_src/asterisk/apps/apps-meetme-cbmysql.txt If you run patch from within the apps directory, you will need to use -p2. Or just cd .. and use the same command as above. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call a url and get a result in the dialplan
Build a script, use curl or wget parse output and use the variable to trigger events either via gotos of via the agi-script you wrote. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matias G. Sent: Friday, March 18, 2005 11:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] call a url and get a result in the dialplan Hi, can a call a php script wich is located in a remote server, someting like calling www.theserveraddress.com/scripts/validate?code=234234swq and get the result which this script generates (a 0 or a 1) back in the dial plan in a direct way or should I create a script which in turn does this? I'm using * CVS HEAD. Also I searched for this for I while but didn't manage to find anything but SendUrl and PHP AGI thanks M. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reply a post
I think you just did -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanishka Somaratne Sent: Friday, March 18, 2005 11:15 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] reply a post Hi how do i reply a question asked in this mailling list. tks Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Optional URL in App. Queue
I have googled for days abt this so finally i turn to the list that knows all :) I want to use the Optional URL parameter in the Queue application to redirect agents to a webpage. Now I cant seem to find a IAX client for windows that supports it. I dont mind a SIP client either but I dont want to install AST-CC or something along those lines on the Asterisk box. Just looking for a simple IAX client which can open up mozilla and show me the url -- regards Vikram (http://www.vicramresearch.com) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime - how to reload ?
Ronald Wiplinger wrote: I had the impression that the command: *CLI realtime load sippeers name 621 (The new configuration was displayed after that command) would re-load the config of phone 621 I may be wrong but realtime load is simply a debug tool for CLI so that you can view what is in the database. You can use realtime update to change info. If you are using RTCaching then (I'm guessing) that you need to wait for the cache period to expire before those values take affect. If you are not using RTC, then the changes should be instant. I think I will look into adding realtime refresh family or something like that. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Group Ring after Timeout
Blibs are IRQ conflicts. cat /proc/interrupts see if the IRQ of the X100P (wcfxo) is being shared with anything. If it is try changing PCI slots - Original Message - From: Reuben Grech To: asterisk-users@lists.digium.com Sent: Friday, March 18, 2005 9:59 AM Subject: [Asterisk-Users] Group Ring after Timeout Dear All, I am listening to blips during conversations when I have an incoming call from an X100P card. This does not happen on all conversations. Any clues? :) ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Problem = Segmentation faults Seemsto be asterisk
Jose R. Ortiz Ubarri wrote: Debugging the code and as you can see in the backtrace the problem is that it is receiving a Null variable (name) and then making the comparison. Is it an asterisk bug? What asterisk should do if the variable name received is NULL? Our chan_sip.c are still not synch'd. I can't help if I don't have the right line numbers. What version of chan_sip are you using? Check inside CVS/Entries and I'll make sure I have the same ver. Then do a make clean; make and reinstall and produce the crash and send the backtrace again. If you can also send the relevant entries in your database that relate to this SIP user. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Last guy to get BV working outbound?
Brian,
You will need to add the following to your broadvoice peer:
user=phone
insecure=very
dtmf=inband
For more info check out:
http://geekgazette.com/index.php?option=com_contenttask=viewid=20Itemid=26
Hope this helps. -john
Brian G wrote:
I have tried everything to get BV working outbound. All worked fine
until the BV change last week. I called BV and they changed me to sip
gen with a new password. I stripped my Asterisk server to one phone on
Zap/1 until I get this working. The same BV account works fine with a
SPA-3000 so I don't suspect a firewall problem.
Symptoms: Asterisk registers with BV Ok
Incoming calls work
Outbound calls send Invite, receive 100, then 401
Asterisk sends an ACK instead of another Invite with credentials
If anyone knows what specifically makes Asterisk respond to the 401 with
credentials for an authenticated Invite, I'd appreciate it. I can't
seem to find this out.
Thanks in advance,
Brian
Here is my sip.conf:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
srvlookup = yes ; Enable DNS SRV lookups on outbound
calls
disallow=all; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
;
; Configuration for BroadVoice
;
register =
[EMAIL PROTECTED]:pword:[EMAIL PROTECTED]
;
[broadvoice]
type=peer
host=sip.broadvoice.com
secret=pword
fromuser=508XXX
username=508XXX
authuser=508XXX
fromdomain=sip.broadvoice.com
context=incoming
canreinvite=no
dtmfmode=inband
qualify=yes
in extensions.conf:
[default]
exten = _81XX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _81XX,2,Congestion()
exten = _81XX,102,busy()
Other Asterisk info:
*CLI sip show registry
Host Username Refresh State
147.135.0.128:5060508XXX 120 Registered
*CLI
*CLI show version
Asterisk CVS-03/11/05-16:07:49 built by [EMAIL PROTECTED] on a i686
running Linux
*CLI
*CLI Mar 17 10:35:08 NOTICE[-245486672]: chan_sip.c:5047
handle_response: Failed to authenticate on INVITE to 'Analog1
sip:[EMAIL PROTECTED];tag=as212bf17
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Re: [Asterisk-Users] Group Ring after Timeout
On Fri, 18 Mar 2005 16:59:36 +0100, Reuben Grech [EMAIL PROTECTED] wrote: Dear All, I am listening to blips during conversations when I have an incoming call from an X100P card. This does not happen on all conversations. Any clues? :) Check cat /proc/interrupts make sure the X100P has it's own interrupt I suspect not. Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Optional URL in App. Queue
Try DIAX. http://www.laser.com/dante/ Vikram Rangnekar wrote: I have googled for days abt this so finally i turn to the list that knows all :) I want to use the Optional URL parameter in the Queue application to redirect agents to a webpage. Now I cant seem to find a IAX client for windows that supports it. I dont mind a SIP client either but I dont want to install AST-CC or something along those lines on the Asterisk box. Just looking for a simple IAX client which can open up mozilla and show me the url ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for quality inbound DID - IAX providers, UK, USA, Australia
Hi All, I am looking for a provider/s of inbound DID IAX numbers, for UK, USA, and Australia. Preferably free or low cost J Can anyone make a good reference? Many thanks Chris PS: I appreciate this is perhaps a little OT, please feel free to reply off list. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with Understanding CODEC
Hello All, Need some small help understanding the CODEC part of Asterisk and my endpoints. All my endpoints are Snom IP Phone and they come with G711a/u and G729A. So far I got mix messages from people that if your endpoints got G729A or G723 codec already, I don't need to buy any licensing for the CODECS. But my question is What about the server it self? Because if I set my priorities in the sip.conf file to use G723, G729 and rest disallow=all, the voice stops working. So in short I do need licensing for G723 and G729 to be installed on the Asterisk box it self? And how many do I need? Please help! Neel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] seg fault when accessing voicemail via any IAX softphone (diax, iax phone)
hello all, i am happy to be part of the asterisk community. i have sucessfully configured my asteisk server with the following (fxo card, fwd-ipkall, fwdout,) everything works great except when i attempt to check my voicemail via any iax softphones for windows (diax, iax phone). basically whenever i attempt to access voicemail (via demo extension 8500) i get a segmentation fault. when it does work there is a distinct dong noise and distortion which occurs when Comedian Mail is announced. this problem does not manifest itself when handling iax calls coming from fwd network. it fails/seg faults 95% of the time. i have copied information from my console, confs, and gdb. *this does not happen with sip clients. nor does it happen with incoming iax calls (like from fwd).* it only seems to happen with local, registered, iax softphones. i am considering purchasing an iaxy but am afraid that this will continue to happen. i searched but could not find anything adressing this. any help would be appreciated. running fedora3 core. my asterisk version (built from cvs): Asterisk CVS-HEAD-03/03/05-16:05:19 built by [EMAIL PROTECTED] on a i686 running Linux here is an entry in my iax.conf for my softphone: [amarfresh] username=amarfresh type=friend auth=md5 host=dynamic context=default secret=abcdefg regexten=6969 mailbox=6969; Notify about mailbox 6969 callerid=Some Host (732) 999- asterisk console crash message: Asterisk Ready. *CLI -- Accepting AUTHENTICATED call from 192.168.4.100: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (gsm|ulaw), priority = mine -- Executing VoiceMailMain(IAX2/[EMAIL PROTECTED], ) in new stack Segmentation fault (core dumped) gdb backtrace: (gdb) bt #0 adsi_transmit_message (chan=0x88ff450, msg=0xf697fe50 \216\005, msglen=9, msgtype=-157811232) at res_adsi.c:338 #1 0xf6f3a113 in adsi_load_session (chan=0x88ff450, app=0xf69c23e4 , ver=1, data=1) at res_adsi.c:949 #2 0xf69b4b61 in vm_authenticate (chan=0x88ff450, mailbox=0xf69813f0 , mailbox_size=80, res_vmu=0xf697fde0, context=0x0, prefix=0xf6981e80 , skipuser=0, maxlogins=3) at app_voicemail.c:2439 #3 0xf69bbe97 in vm_execmain (chan=0x88ff450, data=0x0) at app_voicemail.c:4529 #4 0x08085e66 in pbx_extension_helper (c=0x88ff450, con=0x0, context=0x88ff598 default, exten=0x88ff68c 8500, priority=1, label=0x0, callerid=0x88e1f50 732999, action=0) at pbx.c:501 #5 0x08086f6b in ast_pbx_run (c=0x88ff450) at pbx.c:1917 #6 0x08088741 in pbx_thread (data=0xf697fde0) at pbx.c:2162 #7 0x0080e3ae in start_thread () from /lib/tls/libpthread.so.0 #8 0x0067bb6e in clone () from /lib/tls/libc.so.6 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie question
I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts on it. They seem to run fine inside my network, so that's OK. Now, I want to start using a X100P to connect it to my phone line, to make call routing between internal SIP phones/softphones, my local phoneline and an external SIP server. How do I enable and configure the X100P? I ran the configuration tool locally on the machine (the genzaptelconf thing) and it added a line to the config. Now using the number it gave me, in the trunk config in AMP, I still cannot get an outside line (connected it to a simple analogue pbx system) and call outside the *-server.. Could anyone help me with this? Thanks guys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some IAX questions
Hi All, I am trying to figure out how Asterisk determines which [user] an incoming IAX connection is for? Is it based on their source address? (I think possible) Is it based on their credentials (unlikely, I think, since we can set insecure=very) Also, for a [peer] section - when is the host= resolved? Is name resolution attempted every time the channel is opened? Thanks! Tim (Oh and sorry for the oracle crack yesterday - I couldn't resist ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reply a post
Click on Reply in your mail client, type message, click Send. On Fri, 18 Mar 2005 16:15:24 -, Kanishka Somaratne [EMAIL PROTECTED] wrote: Hi how do i reply a question asked in this mailling list. tks Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voice getting cutoff
What do you think? CPU0 0: 16148159 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 10: 161351663 XT-PIC usb-uhci, wcfxo 11:1276097 XT-PIC usb-uhci, eth0 12: 161350551 XT-PIC ehci-hcd, PS/2 Mouse, wcfxo 14: 138574 XT-PIC ide0 15: 33 XT-PIC ide1 NMI: 0 ERR: 0 Any problems here? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito Sent: Martes, 15 de Marzo de 2005 10:55 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voice getting cutoff check for interrupt conflicts, cat /proc/interrupts - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 15, 2005 8:26 PM Subject: [Asterisk-Users] Voice getting cutoff Guys.. I just noticed that my grandstream handytone 286 ata are having problems with voice cutoffs... We can listen to the person on the zap channel (x100p cards) without problems but they sometimes listen to us with cutoffs.. like He ...lo. ow...r.. you and it comes and goes.. this doesnt happen all the time but often enough. Any ideas what might be happening or what do I need to send you to help me debug this issue... Thx Guys! BTW, my ATAs are using ulaw. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie question
Contact me offlist and I will gice you some info W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bram Sent: Friday, March 18, 2005 10:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie question I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts on it. They seem to run fine inside my network, so that's OK. Now, I want to start using a X100P to connect it to my phone line, to make call routing between internal SIP phones/softphones, my local phoneline and an external SIP server. How do I enable and configure the X100P? I ran the configuration tool locally on the machine (the genzaptelconf thing) and it added a line to the config. Now using the number it gave me, in the trunk config in AMP, I still cannot get an outside line (connected it to a simple analogue pbx system) and call outside the *-server.. Could anyone help me with this? Thanks guys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Manager Interface
I am trying to handle parked calls through the manager interface, but having a lot of trouble, if I try to Park a call I try with this command: Action: Redirect Channel: SIP/211 Exten: 700 Context: phones CallerID: tgj Priority: 1 But it just doesn't work because I have no extension 700 in my dial plan (700 is what I have configured parkedcalls to be in features.conf) I have configured 'blind transfer' to be '#1' in features.conf so I have also tried: Action: Redirect Channel: SIP/211 Exten: #1700 Context: phones CallerID: tgj Priority: 1 But it doesn't work. Now I try to pick up a parked call which has been assigned to 701 like this: Action: Originate Channel: SIP/211 Exten: 701 Context: phones CallerID: tgj Priority: 1 ..but the same problem it just doesn't work - What am I missing? Please help. I might mention that everything works fine using the hard phones. Kind regards Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN Voicemail
This is probably a stupid question. How do I login to voicemail from the PSTN? I can dial *98 from within the system, but when dialing from the PSTN I have it set up to ring a dial group, then to an extensions vmail. During the extensions vmail prompts, I dial *98 and it sends me to the directory. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
