[asterisk-users] Ast11: How to see call progress like in Ast = 1.8
Hi, I just did a test install of Ast 11, and have trouble getting the same logging information that Ast 1.x provided. I'm looking specifically for the logging around call progress / dialplan actions. In ASt 11 I've done the same thing that I did before: core set verbose 60 I also tried overwriting the logger.conf with the distribution one from Ast 11, and setting option logger set level verbose on (never did that on older versions, but was wondering if that would make a difference). Still no joy, Googling around for an answer I did see a changelog with an example of the Call Identifier that shows a detailed logline (of level verbose, something I don't get in 11). ButI've been unable to find an answer. Any hints/tips, I must be overlooking something basic.. TX! Bas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Dear Randulo, Thanks for your suggention. Now i am able to communicate between 2 computers. Regards, Baskar --- randulo [EMAIL PROTECTED] wrote: On Mon, May 19, 2008 at 8:44 AM, bas karan [EMAIL PROTECTED] wrote: [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 handle_request_invite: Call from 'Phone3' to extension '5' rejected because extension not found. -- Registered SIP 'Phone3' at 192.168.1.101 port Extension.conf enteries are, exten = 3,1,Dial(SIP/Phone3,30,tr) exten = 4,1,Dial(SIP/Phone4,30,tr) exten = 5,1,Dial(SIP/Phone5,30,tr) Where is the [sip] context named in the phones context= statement ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Has your work life balance shifted? Find out - http://in.search.yahoo.com/search?fr=na_onnetwork_mail_taglinesei=UTF-8rd=r1p=work+life+balance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension not found
Dear Friends, This is Baskar from Chennai, trying to configure asterisk. Now I planned to start with communication between 2 systems using soft phones. When I tried to call the other computer I am getting the following error message on asterisk terminal, Connected to Asterisk 1.4.18 currently running on asterisker (pid = 2478) Verbosity is at least 3 [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 handle_request_invite: Call from 'Phone3' to extension '5' rejected because extension not found. -- Registered SIP 'Phone3' at 192.168.1.101 port 5060 expires 70 asterisker*CLI SIP.conf Entries are as follows: [Phone3] type = friend secret=Phone3 host = dynamic defaultip = 192.168.1.101 dtmfmode = rfc2833 context = sip callerid = Phone3 3 [Phone4] type = friend secret=Phone4 host = dynamic defaultip = 127.0.0.1 dtmfmode = rfc2833 context = sip callerid = Phone4 4 [Phone5] type = friend secret=Phone5 host = dynamic defaultip = 192.168.1.51 dtmfmode = rfc2833 context = sip callerid = Phone5 5 Extension.conf enteries are, exten = 3,1,Dial(SIP/Phone3,30,tr) exten = 4,1,Dial(SIP/Phone4,30,tr) exten = 5,1,Dial(SIP/Phone5,30,tr) Please help me to fix this issue. Thank in advance. Regards, Baskar Bollywood, fun, friendship, sports and more. You name it, we have it on http://in.promos.yahoo.com/groups/bestofyahoo/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension not found
Dear Randulo, Thanks for your replay. I am new to this concept, Could you explain me little bit extra please? Thanks Regards, Baskar --- randulo [EMAIL PROTECTED] wrote: On Mon, May 19, 2008 at 8:44 AM, bas karan [EMAIL PROTECTED] wrote: [May 19 12:02:29] NOTICE[2559]: chan_sip.c:13879 handle_request_invite: Call from 'Phone3' to extension '5' rejected because extension not found. -- Registered SIP 'Phone3' at 192.168.1.101 port Extension.conf enteries are, exten = 3,1,Dial(SIP/Phone3,30,tr) exten = 4,1,Dial(SIP/Phone4,30,tr) exten = 5,1,Dial(SIP/Phone5,30,tr) Where is the [sip] context named in the phones context= statement ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Meet people who discuss and share your passions. Go to http://in.promos.yahoo.com/groups/bestofyahoo/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
Peter, I have 600's that are 12V 1.5A, + in the center. This differs from some of the other answers, maybe those differences are regional (although that would seem rather silly). HTH B Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
I haven't read trough the thread well enough. The 600 is 12V 1.5A indeed. Too bad they don't all have the same voltage. LST wrote: The IP600 is 12v!!! I fried a 600 when I used power adapter from 601. On 1/3/07, *Alvin Austin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: FWIW, our Polycom IP601 phones use a transformer with output: 24VDC 500mA (center contact is positive). A Polycom reseller (or Polycom sales) could probably give you information on these other two models. Alvin Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 2801 and asterisk
Hello, Is anybody using a Cisco 2801 to connect to the PSTN? I am having some issues regarding SIP responses from the router to asterisk (too general e.g. SIP 404 errors when they should be more specific). Regards, Bas pgpj8PZe7U89B.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help on Music on Hold
Hi, Do you have vad disabled in your dial-peer voice XX voip dial-peer? What kind of MOH are you using; asterisk native or an external player like mpg123? --basv On Fri, Nov 17, 2006 at 08:41:49AM -0500, gc wrote: I am testing asterisk (version 1.2.12.1) on a Dell 1950 server and have this strange problem on music on hold. When I called into a queue using SIP from PSTN line which goes through our cisco gateway (cisco 5300), asterisk will start play music on hold. But this MOH seems at voice activation mode. That is only when I make noice on my end then I can hear music otherwise I will hear silence. I have another asterisk (version 1.2.9.1) running on an older Dell server and MOH works fine for call from PSTN. So my guess is that maybe there is some settings in asterisk cause this problem. Any suggestion about this problem? GG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users pgpOMULkKCmYV.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to cross compile asterisk for Axis ETRAX 100LX foxboard embedded device on Debian
Hi everybody, I bought a foxboard an embedded device with an axis processor, I'd like to cross-compile Asterisk for the foxboard on my Debian box. I use a software development kit from Axis and I have a little tutorial from the board manufacturer on how to cross compile a little hello world program for the board http://www.acmesystems.it/index.php/How_to_compile_a_C_application Here are the programms needed for axis cross compilation found on this page http://www.acmesystems.it/index.php/Installing_the_Axis_SDK pmake_1.98-3_i386.deb cris-dist_1.63-1_i386.deb devboard-R2_01.tar.gz devboard-R2_01-distfiles.tar.gz The problem is that I don't know what to edit in the asterisk Makefile: # If cross compiling, define these to suit # CROSS_COMPILE=/opt/montavista/pro/devkit/arm/xscale_be/bin/xscale_be- # CROSS_COMPILE_BIN=/opt/montavista/pro/devkit/arm/xscale_be/bin/ # CROSS_COMPILE_TARGET=/opt/montavista/pro/devkit/arm/xscale_be/target Here a some paths I have on my computer: [EMAIL PROTECTED]:/home/francois/foxboard# find . -iname 'bin' ./devboard-R2_01/tools/build-R2_12_4/bin ./devboard-R2_01/target/cris-axis-linux-gnu/usr/bin ./devboard-R2_01/target/cris-axis-linux-gnu/bin [EMAIL PROTECTED]:/home/francois/foxboard# find . -iname 'target' ./devboard-R2_01/os/linux-2.6-tag--devboard-R2_01/include/config/ip/nf/target ./devboard-R2_01/target Can someone explain me what I should change in Asterisk Makefile, -- Francois-Xavier Bas RSS-Global Technologies Ltd. Bachemer Strasse 266 50935 Cologne Germany phone: +49221 297-6491 email: [EMAIL PROTECTED] url: www.rss-global.com begin:vcard fn:Francois Bas n:Bas;Francois org:RSS Global Technologies Ltd. adr:;;Bachemer Strasse 266;Cologne;;50935;Germany email;internet:[EMAIL PROTECTED] title:System Administrator tel;work:+49 221 2976 491 x-mozilla-html:TRUE url:http://www.rss-global.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound sipgate number forwarding to differnet users
How can I forward my offcial sipgate number to different users, I would like to know if it is possible to append a local user number to my official number when dialing, then in this way it could be forwarded using the suffixe local user number.The prefixe number would be the official sipgate number.Are tehre some companies that provide this kind of service instead of suscribing to many official numbers? -- Francois-Xavier Bas RSS-Global Technologies Ltd. Bachemer Strasse 266 50935 Cologne Germany phone: +49221 297-6491 email: [EMAIL PROTECTED] url:www.rss-global.com begin:vcard fn:Francois Bas n:Bas;Francois org:RSS Global Technologies Ltd. adr:;;Bachemer Strasse 266;Cologne;;50935;Germany email;internet:[EMAIL PROTECTED] title:System Administrator tel;work:+49 221 2976 491 x-mozilla-html:TRUE url:http://www.rss-global.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to forward inbound sipgate calls to different users in my entreprise, (
Hi everybody, I would like to know if it is possible to add a suffixe number to my official sipgate number? For example my official sipgate number is 123456 My local users numbers are 0001 0002 Could it be possible to compose from internet 1234560001 or 1234560002 , then the message could be forwarded to user 0001 or 0002. Actually I can only forward my sipgate number to only one user. So I am looking for a service or a solution that would allow me to forward internet calls to different users. Because now to do this, I would have to suscribe for many official accounts , in this way I could forward each account for each user, but it wouldn't be practice. So I hope there is a solution for that. Thanks a lot I wait some feedback. -- Francois-Xavier Bas RSS-Global Technologies Ltd. Bachemer Strasse 266 50935 Cologne Germany phone: +49221 297-6491 email: [EMAIL PROTECTED] url:www.rss-global.com begin:vcard fn:Francois Bas n:Bas;Francois org:RSS Global Technologies Ltd. adr:;;Bachemer Strasse 266;Cologne;;50935;Germany email;internet:[EMAIL PROTECTED] title:System Administrator tel;work:+49 221 2976 491 x-mozilla-html:TRUE url:http://www.rss-global.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lingo(.com) and Asterisk
Hello, A long Google search didn't turn any clear answer. Does somebody use Asterisk in combination with Lingo? Thank you, Bas ___ 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] Using * @ Home, all seems to work, but no sound to Softphone
Hello,
To do some testing with Asterisk installed the latest Asterisk @ Home in a
Vmware system. All worked fine, I can access the web interface (AMP). I have
setup the extention and X-Lite softphone according to the description in the
Wike (http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite).
I can dial 200 (the softphone extention) and 1234 and they connect (the
softphone shows this, as well as the call record), but I don't get any sound
from it. I would expect to hear the Festival output that the asterisk
console shows it is generating.
I tried both X_lite and Firefly softphone, but phones do give me sounds when
pressing buttons etc, so it's not my loudspeaker ;-)
Any suggestions on what might be wrong?
Bas
Attached is the relevant output from the debug log:
Mar 29 02:46:11 WARNING[1433]: Inband DTMF is not supported on codec gsm.
Use RFC2833
Mar 29 02:46:11 DEBUG[1433]: Scheduling timer at 0 sample intervals
Mar 29 02:46:11 VERBOSE[1433]: == Spawn extension (macro-exten-vm, novm, 3)
exited non-zero on 'SIP/200-a5f0' in macro 'exten-vm'
Mar 29 02:46:11 VERBOSE[1433]: == Spawn extension (from-internal, 200, 1)
exited non-zero on 'SIP/200-a5f0'
Mar 29 02:46:11 VERBOSE[1433]: -- Executing
[1;36;40mMacro[0;37;40m([1;35;40mSIP/200-a5f0[0;37;40m,
[1;35;40mhangupcall[0;37;40m) in new stack
Mar 29 02:46:12 VERBOSE[1433]: -- Executing
[1;36;40mResetCDR[0;37;40m([1;35;40mSIP/200-a5f0[0;37;40m,
[1;35;40mw[0;37;40m) in new stack
Mar 29 02:46:12 DEBUG[1433]: cdr_mysql: inserting a CDR record.
Mar 29 02:46:12 DEBUG[1433]: cdr_mysql: SQL command as follows: INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-03-29
02:46:03','\Bas Rijniersce\ 200','200','200','from-internal',
'SIP/200-a5f0','','ResetCDR','w',9,8,'ANSWERED',3,'')
Mar 29 02:46:12 VERBOSE[1433]: -- Executing
[1;36;40mNoCDR[0;37;40m([1;35;40mSIP/200-a5f0[0;37;40m,
[1;35;40m[0;37;40m) in new stack
Mar 29 02:46:12 WARNING[1433]: CDR on channel 'SIP/200-a5f0' not posted
Mar 29 02:46:12 WARNING[1433]: CDR on channel 'SIP/200-a5f0' lacks end
Mar 29 02:46:12 VERBOSE[1433]: -- Executing
[1;36;40mWait[0;37;40m([1;35;40mSIP/200-a5f0[0;37;40m,
[1;35;40m5[0;37;40m) in new stack
Mar 29 02:46:12 VERBOSE[1433]: == Spawn extension (macro-hangupcall, s, 3)
exited non-zero on 'SIP/200-a5f0' in macro 'hangupcall'
Mar 29 02:46:12 VERBOSE[1433]: == Spawn extension (from-internal, h, 1)
exited non-zero on 'SIP/200-a5f0'
Mar 29 02:46:12 DEBUG[1433]: update_user_counter(200) - decrement inUse
counter
Mar 29 02:46:13 DEBUG[1433]: Auto destroying call
'[EMAIL PROTECTED]'
Mar 29 02:46:15 DEBUG[1433]: Setting NAT on RTP to 0
Mar 29 02:46:15 DEBUG[1433]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 1: Found
Mar 29 02:46:15 DEBUG[1433]: Setting NAT on RTP to 0
Mar 29 02:46:15 DEBUG[1433]: Check for res for 200
Mar 29 02:46:15 DEBUG[1433]: Call from user '200' is 1 out of 0
Mar 29 02:46:15 DEBUG[1433]: build_route: Contact hop:
Mar 29 02:46:15 VERBOSE[1433]: -- Executing
[1;36;40mAnswer[0;37;40m([1;35;40mSIP/200-74a5[0;37;40m,
[1;35;40m[0;37;40m) in new stack
Mar 29 02:46:15 VERBOSE[1433]: -- Executing
[1;36;40mAGI[0;37;40m([1;35;40mSIP/200-74a5[0;37;40m,
[1;35;40mfestival-script.pl|Welcome to the wonderful world of Asterisk!
Your phone number is 200.[0;37;40m) in new stack
Mar 29 02:46:15 VERBOSE[1433]: -- Launched AGI Script
/var/lib/asterisk/agi-bin/festival-script.pl
I also tried without forcing the gsm codec. Didn't make a difference
___
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
