Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]
@ Tzafrir you mean say i shouldn't use "-server" kernel for asterisk ? -Satish Date: Mon, 11 Apr 2011 07:45:01 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199] On Sun, Apr 10, 2011 at 9:12 AM, Tzafrir Cohen wrote: Off-topic: On Fri, Apr 08, 2011 at 03:30:58PM +, satish patel wrote: [snip] > System: Linux/2.6.32-24-server built by root on x86_64 > 2011-03-22 18:38:19 UTC Ubuntu has a separate -server kernel variant. From what I understand, using it is not a good idea on a Asterisk system, as it is intended to an application such as a file server, optimized for higher throughput. Asterisk is closer to a desktop multimedia program, which prefers low latency to high throughput. Is that recommendation still valid? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- Simple answer: RTFM http://manpages.ubuntu.com/manpages/lucid/man7/time.7.html http://ubuntuforums.org/showthread.php?t=1651629 The "purpose" of the distro sets the timer. I am sure there is a workaround for server to use an Asterisk friendly kernel timer. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Asterisk E1 connection
For PRI coross over cable following is pin layout 1 <---> 4 2 <---> 5 > Date: Mon, 11 Apr 2011 10:43:51 -0300 > From: aco1...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk-Asterisk E1 connection > > Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both > boxes. I need to connect both PBXs with E1/R2 and UTP cable. > > What are the requirements to deploy the UTP cable ??? Straight-through > or crossover ??? What are the pinouts in both peers ??? > > Thanks a lot, > > Alejandro > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]
I don't understand what you guys talking about? You mean say there is a issue in ubuntu kernel to use asterisk? -- Sent from my iPhone On Apr 11, 2011, at 8:05 AM, Tzafrir Cohen wrote: On Mon, Apr 11, 2011 at 07:45:01AM -0400, Steve Totaro wrote: On Sun, Apr 10, 2011 at 9:12 AM, Tzafrir Cohen >wrote: Off-topic: On Fri, Apr 08, 2011 at 03:30:58PM +, satish patel wrote: [snip] System: Linux/2.6.32-24-server built by root on x86_64 2011-03-22 18:38:19 UTC Ubuntu has a separate -server kernel variant. From what I understand, using it is not a good idea on a Asterisk system, as it is intended to an application such as a file server, optimized for higher throughput. Asterisk is closer to a desktop multimedia program, which prefers low latency to high throughput. Is that recommendation still valid? Simple answer: RTFM http://manpages.ubuntu.com/manpages/lucid/man7/time.7.html http://ubuntuforums.org/showthread.php?t=1651629 The "purpose" of the distro sets the timer. I am sure there is a workaround for server to use an Asterisk friendly kernel timer. Sure. There's e.g. a -preempt kernel variant ( http://packages.ubuntu.com/lucid/linux-image-2.6.32-24-preempt ). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I grab via svn client and source you gave me. Can you fix original brach ? -- Sent from my iPhone On Apr 10, 2011, at 11:51 AM, Paul Belanger wrote: On 11-04-10 09:14 AM, Tzafrir Cohen wrote: On Fri, Apr 08, 2011 at 06:10:21PM +, satish patel wrote: I tried to compile your version and got bunch of error on "make" and it failed to compile. root@satish-desktop:/home/satish/issue18183# make How did you get that code? It is from a branch I created a few months back, and I have not looked at it in a while. That said, there maybe issues with it. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
Bump up! Please help here -- Sent from my iPhone On Apr 8, 2011, at 2:10 PM, satish patel wrote: I tried to compile your version and got bunch of error on "make" and it failed to compile. root@satish-desktop:/home/satish/issue18183# make [CC] chan_iax2.c -> chan_iax2.o chan_iax2.c: In function âsocket_processâ: chan_iax2.c:11533: error: invalid storage class for function âiax2_p rocess_thread_cleanupâ chan_iax2.c:11532: warning: no previous prototype for âiax2_process_ thread_cleanupâ chan_iax2.c:11544: error: invalid storage class for function âiax2_p rocess_threadâ chan_iax2.c:11543: warning: no previous prototype for âiax2_process_ threadâ chan_iax2.c:11683: error: invalid storage class for function âiax2_d o_registerâ chan_iax2.c:11682: warning: no previous prototype for âiax2_do_regis terâ chan_iax2.c:11744: error: invalid storage class for function âiax2_p rovisionâ chan_iax2.c:11743: warning: no previous prototype for âiax2_provisio nâ chan_iax2.c:11796: error: invalid storage class for function âiax2_p rov_appâ chan_iax2.c:11795: warning: no previous prototype for âiax2_prov_ap pâ chan_iax2.c:11825: error: invalid storage class for function âhandle _cli_iax2_provisionâ chan_iax2.c:11824: warning: no previous prototype for âhandle_cli_ia x2_provisionâ chan_iax2.c:11864: error: invalid storage class for function â__iax2 _poke_noanswerâ chan_iax2.c:11863: warning: no previous prototype for â__iax2_poke_n oanswerâ chan_iax2.c:11887: error: invalid storage class for function âiax2_p oke_noanswerâ ... ... ... chan_iax2.c:14723: warning: no previous prototype for â__reg_moduleâ chan_iax2.c:14723: error: invalid storage class for function â__unre g_moduleâ chan_iax2.c:14723: warning: no previous prototype for â__unreg_modul eâ chan_iax2.c:14723: error: expected declaration or statement at end of input chan_iax2.c:14723: warning: unused variable âast_module_infoâ make[1]: *** [chan_iax2.o] Error 1 make: *** [channels] Error 2 root@satish-desktop:/home/satish/issue18183# > Date: Fri, 8 Apr 2011 13:16:30 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > > On 11-04-08 12:56 PM, Paul Belanger wrote: > > On 11-04-08 11:55 AM, satish patel wrote: > >> > >> @Paul - many time i am gettting following SIP error when channel isn't > >> available. I want to get rid on this revers thing. I tried all version > >> 1.8.1,1.8.2,1.8.3 but not fix :( > >> > > Best you can do is collect a full debug[1] log and see when the issue is > > introduced. > > > > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information > > > Do you mind trying the following branch[2]? Not sure if it will help, > but I made some changes to chan_iax2 a few months ago. > > [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] IAX2/0.0.29.199
I tried to compile your version and got bunch of error on "make" and it failed to compile. root@satish-desktop:/home/satish/issue18183# make [CC] chan_iax2.c -> chan_iax2.o chan_iax2.c: In function âsocket_processâ: chan_iax2.c:11533: error: invalid storage class for function âiax2_process_thread_cleanupâ chan_iax2.c:11532: warning: no previous prototype for âiax2_process_thread_cleanupâ chan_iax2.c:11544: error: invalid storage class for function âiax2_process_threadâ chan_iax2.c:11543: warning: no previous prototype for âiax2_process_threadâ chan_iax2.c:11683: error: invalid storage class for function âiax2_do_registerâ chan_iax2.c:11682: warning: no previous prototype for âiax2_do_registerâ chan_iax2.c:11744: error: invalid storage class for function âiax2_provisionâ chan_iax2.c:11743: warning: no previous prototype for âiax2_provisionâ chan_iax2.c:11796: error: invalid storage class for function âiax2_prov_appâ chan_iax2.c:11795: warning: no previous prototype for âiax2_prov_appâ chan_iax2.c:11825: error: invalid storage class for function âhandle_cli_iax2_provisionâ chan_iax2.c:11824: warning: no previous prototype for âhandle_cli_iax2_provisionâ chan_iax2.c:11864: error: invalid storage class for function â__iax2_poke_noanswerâ chan_iax2.c:11863: warning: no previous prototype for â__iax2_poke_noanswerâ chan_iax2.c:11887: error: invalid storage class for function âiax2_poke_noanswerâ ... ... ... chan_iax2.c:14723: warning: no previous prototype for â__reg_moduleâ chan_iax2.c:14723: error: invalid storage class for function â__unreg_moduleâ chan_iax2.c:14723: warning: no previous prototype for â__unreg_moduleâ chan_iax2.c:14723: error: expected declaration or statement at end of input chan_iax2.c:14723: warning: unused variable âast_module_infoâ make[1]: *** [chan_iax2.o] Error 1 make: *** [channels] Error 2 root@satish-desktop:/home/satish/issue18183# > Date: Fri, 8 Apr 2011 13:16:30 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > > On 11-04-08 12:56 PM, Paul Belanger wrote: > > On 11-04-08 11:55 AM, satish patel wrote: > >> > >> @Paul - many time i am gettting following SIP error when channel isn't > >> available. I want to get rid on this revers thing. I tried all version > >> 1.8.1,1.8.2,1.8.3 but not fix :( > >> > > Best you can do is collect a full debug[1] log and see when the issue is > > introduced. > > > > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information > > > Do you mind trying the following branch[2]? Not sure if it will help, > but I made some changes to chan_iax2 a few months ago. > > [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I have just compiled asterisk 1.6.x and its working without any issue no error related revers lookup etc.. Look like there is some glitch in asterisk 1.8 :( -S > Date: Fri, 8 Apr 2011 13:16:30 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > > On 11-04-08 12:56 PM, Paul Belanger wrote: > > On 11-04-08 11:55 AM, satish patel wrote: > >> > >> @Paul - many time i am gettting following SIP error when channel isn't > >> available. I want to get rid on this revers thing. I tried all version > >> 1.8.1,1.8.2,1.8.3 but not fix :( > >> > > Best you can do is collect a full debug[1] log and see when the issue is > > introduced. > > > > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information > > > Do you mind trying the following branch[2]? Not sure if it will help, > but I made some changes to chan_iax2 a few months ago. > > [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I have opened case here: https://issues.asterisk.org/view.php?id=19087 > Date: Fri, 8 Apr 2011 13:16:30 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > > On 11-04-08 12:56 PM, Paul Belanger wrote: > > On 11-04-08 11:55 AM, satish patel wrote: > >> > >> @Paul - many time i am gettting following SIP error when channel isn't > >> available. I want to get rid on this revers thing. I tried all version > >> 1.8.1,1.8.2,1.8.3 but not fix :( > >> > > Best you can do is collect a full debug[1] log and see when the issue is > > introduced. > > > > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information > > > Do you mind trying the following branch[2]? Not sure if it will help, > but I made some changes to chan_iax2 a few months ago. > > [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2/0.0.29.199
I can try but i have same issue with chan_sip channel also. and next we have scheduled to put this box 1.8.3.2 in production :( -S > Date: Fri, 8 Apr 2011 13:16:30 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] IAX2/0.0.29.199 > > On 11-04-08 12:56 PM, Paul Belanger wrote: > > On 11-04-08 11:55 AM, satish patel wrote: > >> > >> @Paul - many time i am gettting following SIP error when channel isn't > >> available. I want to get rid on this revers thing. I tried all version > >> 1.8.1,1.8.2,1.8.3 but not fix :( > >> > > Best you can do is collect a full debug[1] log and see when the issue is > > introduced. > > > > [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information > > > Do you mind trying the following branch[2]? Not sure if it will help, > but I made some changes to chan_iax2 a few months ago. > > [2] http://svn.asterisk.org/svn/asterisk/team/pabelanger/issue18183/ > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk login to voicemail
I have this for same function.
[voice-mail]
;VM for external users calling from PSTN prompt for mailbox number and pin
exten => 8000,1,Answer()
exten => 8000,n,Wait(1)
exten => 8000,n,VoicemailMain(@default)
exten => 8000,n,Hangup()
;VM for internal users only pin
exten => 8500,1,Answer()
exten => 8500,n,Wait(1)
exten => 8500,n,VoicemailMain(${CALLERID(num):-4}@default)
exten => 8500,n,Hangup()
exten => i,1,playback(invalid)
exten => i,2,hangup
Date: Fri, 8 Apr 2011 12:26:27 -0400
From: vipki...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk login to voicemail
can you explain how this can be done simpler?
On Fri, Apr 8, 2011 at 10:10 AM, Satish Patel wrote:
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa wrote:
Wow, thanks, that worked...in case anyone is interested this is what i did
[voicemail]exten => a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI->set_variable("MAILBOXID",
$options);$AGI->set_variable("MAILBOXCONTEXT","4");$AGI->set_context("voicemail");
$AGI->exec("VoiceMail", $options);
now the question is how to I get the VoiceMailMain to not ask for "Mailbox" and
already know which mailbox and just prompt for "Password"
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo
wrote:
> Unfortunately, that solution will not work for me... The user must be able to
> hit * during the greeting of any voicemail and be prompted for the "Password"
> to that particular mailbox looks like i got a lot of programming to do to
> create a work around for this... thanks for your help...
Forgive me if i'm wrong, but you guys seem to be over complicating things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail context.
Example:
Exten => a, 1, VoicemailMain(@default)
Exten => a, 2, Hangup
When using the star '*' it's important to note that the context you placed the
application voicemail in is irrelevant, it's the context for the voicemail box
that we're looking for in the dialplan for the jump to the 'a' extension.
So this is what i do...
Before passing the call to the voicemail app, i set ${MAILBOXCONTEXT} to the
correct context, and i set ${MAILBOXID} to the mailbox name.
Then, in extensions.conf, I added this:-[voicemail]
exten => a,1,Playback(astcc-please-enter-your)
exten => a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})When the user presses
*, they are passed to the 'a' extension above and into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan JournoKesher Communications (UK)Business Phone Systems | Hosted PBX
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Re: [asterisk-users] IAX2/0.0.29.199
Look at this sip debug its saying something related Retransmitting #1 (no NAT)
to 0.0.29.200:5060:
<>
-- Executing [7624@from-sip:1] Macro("SIP/7527-00c2",
"stdexten,7624,SIP/7624") in new stack
-- Executing [s@macro-stdexten:1] Dial("SIP/7527-00c2",
"SIP/7624&IAX2/7624,20,t") in new stack
== Using SIP RTP CoS mark 5
[Apr 8 12:20:53] WARNING[15194]: acl.c:698 ast_ouraddrfor: Cannot connect
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 0.0.29.200:5060:
INVITE sip:7624 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
Max-Forwards: 70
From: "Cambridge Guest" ;tag=as6f6822ba
To:
Contact:
Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3.2
Date: Fri, 08 Apr 2011 19:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 1407056235 1407056235 IN IP4 172.30.1.47
s=Asterisk PBX 1.8.3.2
c=IN IP4 172.30.1.47
t=0 0
m=audio 16720 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
[Apr 8 12:20:53] WARNING[15194]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2ef3f00 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
-- Called 7624
-- Called 7624
Retransmitting #1 (no NAT) to 0.0.29.200:5060:
INVITE sip:7624 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.47:5060;branch=z9hG4bK3af914e2
Max-Forwards: 70
From: "Cambridge Guest" ;tag=as6f6822ba
To:
Contact:
Call-ID: 0ca7784d38d29be168f8f85711c43e4f@172.30.1.47:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.3.2
Date: Fri, 08 Apr 2011 19:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257
> Date: Fri, 8 Apr 2011 11:12:59 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
>
> On 11-04-08 10:48 AM, satish patel wrote:
> >
> > Where this revers IP comes from ?
> >
> >== Using SIP RTP CoS mark 5
> > -- Executing [7623@from-sip:1] Macro("SIP/7527-006b",
> > "stdexten,7623,SIP/7623") in new stack
> > -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b",
> > "SIP/7623&IAX2/7623,20,t") in new stack
> > -- Hungup 'IAX2/0.0.29.199:4569-5255'
> > -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b",
> > "IAX2/0.0.29.199:4569-5255") in new stack
> > -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in
> > new stack
> > -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'
> >
> Asterisk 1.8? Are you using realtime? Looks to be an issue with
> netsock2.c.
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
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Re: [asterisk-users] IAX2/0.0.29.199
@Paul - many time i am gettting following SIP error when channel isn't
available. I want to get rid on this revers thing. I tried all version
1.8.1,1.8.2,1.8.3 but not fix :(
[Apr 8 11:52:22] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2f40580 (len 793) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 8 11:52:26] NOTICE[13912]: chan_iax2.c:4643 __auto_congest:
Auto-congesting call due to slow response
-Satish
> Date: Fri, 8 Apr 2011 11:12:59 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
>
> On 11-04-08 10:48 AM, satish patel wrote:
> >
> > Where this revers IP comes from ?
> >
> >== Using SIP RTP CoS mark 5
> > -- Executing [7623@from-sip:1] Macro("SIP/7527-006b",
> > "stdexten,7623,SIP/7623") in new stack
> > -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b",
> > "SIP/7623&IAX2/7623,20,t") in new stack
> > -- Hungup 'IAX2/0.0.29.199:4569-5255'
> > -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b",
> > "IAX2/0.0.29.199:4569-5255") in new stack
> > -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in
> > new stack
> > -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'
> >
> Asterisk 1.8? Are you using realtime? Looks to be an issue with
> netsock2.c.
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
> --
> _
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Re: [asterisk-users] IAX2/0.0.29.199
No I am not using any realtime config. its text file..
shirley*CLI> core show settings
PBX Core settings
-
Version: 1.8.3.2
Build Options: LOADABLE_MODULES
Maximum calls: 250 (Current 0)
Maximum open file handles: Not set
Verbosity: 3
Debug level: 0
Maximum load average:0.00
Minimum free memory: 0 MB
Startup time:15:08:59
Last reload time:15:08:59
System: Linux/2.6.32-24-server built by root on x86_64
2011-03-22 18:38:19 UTC
System name:
Entity ID: 00:30:48:77:1c:3c
Default language:en
Language prefix: Enabled
User name and group: asterisk/asterisk
Executable includes: Disabled
Transcode via SLIN: Enabled
Internal timing: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled
* Subsystems
-
Manager (AMI): Enabled
Web Manager (AMI/HTTP): Disabled
Call data records: Enabled
Realtime Architecture (ARA): Disabled
* Directories
-
Configuration file:
Configuration directory: /etc/asterisk
Module directory:/usr/lib/asterisk/modules
Spool directory: /var/spool/asterisk
Log directory: /var/log/asterisk
Run/Sockets directory: /var/run/asterisk
PID file:/var/run/asterisk/asterisk.pid
VarLib directory:/var/lib/asterisk
Data directory: /var/lib/asterisk
ASTDB: /var/lib/asterisk/astdb
IAX2 Keys directory: /var/lib/asterisk/keys
AGI Scripts directory: /var/lib/asterisk/agi-bin
> Date: Fri, 8 Apr 2011 11:12:59 -0400
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] IAX2/0.0.29.199
>
> On 11-04-08 10:48 AM, satish patel wrote:
> >
> > Where this revers IP comes from ?
> >
> >== Using SIP RTP CoS mark 5
> > -- Executing [7623@from-sip:1] Macro("SIP/7527-006b",
> > "stdexten,7623,SIP/7623") in new stack
> > -- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b",
> > "SIP/7623&IAX2/7623,20,t") in new stack
> > -- Hungup 'IAX2/0.0.29.199:4569-5255'
> > -- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b",
> > "IAX2/0.0.29.199:4569-5255") in new stack
> > -- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in
> > new stack
> > -- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'
> >
> Asterisk 1.8? Are you using realtime? Looks to be an issue with
> netsock2.c.
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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[asterisk-users] IAX2/0.0.29.199
Where this revers IP comes from ?
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-006b",
"stdexten,7623,SIP/7623") in new stack
-- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-006b",
"SIP/7623&IAX2/7623,20,t") in new stack
-- Hungup 'IAX2/0.0.29.199:4569-5255'
-- Executing [s@macro-stdexten:2] NoOp("SIP/7527-006b",
"IAX2/0.0.29.199:4569-5255") in new stack
-- Executing [s@macro-stdexten:3] NoOp("SIP/7527-006b", "0&0") in new
stack
-- Auto fallthrough, channel 'SIP/7527-006b' status is 'UNKNOWN'
--
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Re: [asterisk-users] asterisk login to voicemail
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--
Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa wrote:
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten => a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI->set_variable("MAILBOXID", $options);
$AGI->set_variable("MAILBOXCONTEXT","4");
$AGI->set_context("voicemail");
$AGI->exec("VoiceMail", $options);
now the question is how to I get the VoiceMailMain to not ask for
"Mailbox" and already know which mailbox and just prompt for
"Password"
On Thu, Apr 7, 2011 at 6:44 PM, Dan Journo > wrote:
> Unfortunately, that solution will not work for me... The user must
be able to hit * during the greeting of any voicemail and be
prompted for the "Password" to that particular mailbox looks
like i got a lot of programming to do to create a work around for
this... thanks for your help...
Forgive me if i'm wrong, but you guys seem to be over complicating
things.
Taken from: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
during the prompt if the caller presses:
'*' - the call jumps to extension 'a' in the current voicemail
context.
Example:
Exten => a, 1, VoicemailMain(@default)
Exten => a, 2, Hangup
When using the star '*' it's important to note that the context you
placed the application voicemail in is irrelevant, it's the context
for the voicemail box that we're looking for in the dialplan for the
jump to the 'a' extension.
So this is what i do...
Before passing the call to the voicemail app, i set $
{MAILBOXCONTEXT} to the correct context, and i set ${MAILBOXID} to
the mailbox name.
Then, in extensions.conf, I added this:-
[voicemail]
exten => a,1,Playback(astcc-please-enter-your)
exten => a,n,VoicemailMain(${MAILBOXID}@${MAILBOXCONTEXT})
When the user presses *, they are passed to the 'a' extension above
and into VoicemailMain.
I'm sure you can turn this into AGI easily enough if needed.
Dan Journo
Kesher Communications (UK)
Business Phone Systems | Hosted PBX
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Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Yes! You are right! Its working. Now issue is we have SIP extension for
local office users and same number has IAX extension for remote
traveling users. How could i use ChanIsAvail with best action ?
I did following
exten => s,1,ChanIsAvail(${ARG2}&IAX2/${ARG1},20,t)
exten => s,n,NoOp(${AVAILCHAN})
exten => s,n,Set(NewVar=${CUT(AVAILCHAN,,1)})
exten => s,n,NoOp(${NewVar})
exten => s,n,Dial(${NewVar}/${EXTEN})
exten => s,n,Hangup()
And in result i got following: Why its looking at IAX2/0.0.29.199 what is
0.0.29.199?
shirley*CLI>
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-004c",
"stdexten,7623,SIP/7623") in new stack
-- Executing [s@macro-stdexten:1] ChanIsAvail("SIP/7527-004c",
"SIP/7623&IAX2/7623,20,t") in new stack
-- Hungup 'IAX2/0.0.29.199:4569-2707'
-- Executing [s@macro-stdexten:2] NoOp("SIP/7527-004c",
"IAX2/0.0.29.199:4569-2707") in new stack
-- Executing [s@macro-stdexten:3] Set("SIP/7527-004c",
"NewVar=IAX2/0.0.29.199:4569") in new stack
-- Executing [s@macro-stdexten:4] NoOp("SIP/7527-004c",
"IAX2/0.0.29.199:4569") in new stack
-- Executing [s@macro-stdexten:5] Dial("SIP/7527-004c",
"IAX2/0.0.29.199:4569/s") in new stack
-- Called 0.0.29.199:4569/s
[Apr 7 16:59:21] NOTICE[13915]: chan_iax2.c:4643 __auto_congest:
Auto-congesting call due to slow response
-- IAX2/0.0.29.199:4569-3390 is circuit-busy
-- Hungup 'IAX2/0.0.29.199:4569-3390'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-stdexten:6] Hangup("SIP/7527-004c", "") in new
stack
== Spawn extension (macro-stdexten, s, 6) exited non-zero on
'SIP/7527-004c' in macro 'stdexten'
== Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-004c'
> To: asterisk-users@lists.digium.com
> From: isr...@gmail.com
> Date: Thu, 7 Apr 2011 20:49:04 +
> Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
>
> That should be CUT all caps I think
> -Original Message-
> From: satish patel
> Sender: asterisk-users-boun...@lists.digium.com
> Date: Thu, 7 Apr 2011 20:45:21
> To: asterisk-users
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
>
> --
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>
>
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Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
They are on valid IP address range and working properly but when i switched off
that phone and dialing number from other phone i am getting following WARNING!!
So i would like to have some thing like who check CHANNEL first and then say
"Phone is not register" or If phone is available it will ring phone.
I guess ChanIsAvail will fix my issue.
http://www.asteriskguru.com/tutorials/chanisavail_image60455.jpg
But now my asterisk saying i don't have cut application :( How to compile
app_cut.so i didn't find anything related to this in asterisk source.
-- User entered nothing.
[Apr 7 16:36:53] WARNING[14134]: pbx.c:4055 pbx_extension_helper: No
application 'Cut' for extension (macro-stdexten, s, 3)
== Spawn extension (macro-stdexten, s, 3) exited non-zero on
'SIP/7527-003a' in macro 'stdexten'
> Date: Thu, 7 Apr 2011 16:40:12 -0400
> From: p...@dugasenterprises.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
>
> Just a guess but is it possible one of your SIP peers (7623 or 7624)
> has an invalid IP address of 0.0.29.200? I wonder what "sip show
> peers" shows.
>
>
> On Thu, Apr 7, 2011 at 4:03 PM, satish patel wrote:
> >
> > Re-opening this issue.
> >
> > If i dial number which doesn't existing then i am getting following error.
> > So is there anyway i can fix my dialplan to check whether that number exist
> > or not or i can check channel status.
> >
> >
> >
> > shirley*CLI>
> > == Using SIP RTP CoS mark 5
> > -- Executing [7623@from-sip:1] Macro("SIP/7527-0032",
> > "stdexten,7623,sip/7623&sip/7624") in new stack
> > -- Executing [s@macro-stdexten:1] Dial("SIP/7527-0032",
> > "sip/7623&sip/7624&IAX2/7623,20,t") in new stack
> > [Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to
> > create channel of type 'sip' (cause 20 - Unknown)
> > == Using SIP RTP CoS mark 5
> > [Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect
> > [Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
> > -- Called 7624
> > -- Called 7623
> > [Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest:
> > Auto-congesting call due to slow response
> > -- IAX2/0.0.29.199:4569-13525 is circuit-busy
> > -- Hungup 'IAX2/0.0.29.199:4569-13525'
> > [Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt:
> > Retransmission timeout reached on transmission
> > 6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical
> > Request) -- See doc/sip-retransmit.txt.
> > Packet timed out after 32000ms with no response
> > [Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
> > == Spawn extension (macro-stdexten, s, 1) exited non-zero on
> > 'SIP/7527-0032' in macro 'stdexten'
> > == Spawn extension (from-sip, 7623, 1) exited non-zero on
> > 'SIP/7527-0032'
> > [Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
> >
> >
> >
> >
> >
> > From: satish...@hotmail.com
> > To: asterisk-users@lists.digium.com
> > Date: Mon, 4 Apr 2011 20:22:55 +
> > Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
> >
> >
> > Thanks for reply!
> >
> > I found this problem only with X-lite version of softphone. Other phones
> > are working fine without any WARNING! look like X-lite has some short of
> > SIP issue.
> >
> > -S
> >
> >
> >
> >> From: mden...@gmail.com
> >> Date: Mon, 4 Apr 2011 15:59:43 -04
Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Re-opening this issue.
If i dial number which doesn't existing then i am getting following error. So
is there anyway i can fix my dialplan to check whether that number exist or not
or i can check channel status.
shirley*CLI>
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-0032",
"stdexten,7623,sip/7623&sip/7624") in new stack
-- Executing [s@macro-stdexten:1] Dial("SIP/7527-0032",
"sip/7623&sip/7624&IAX2/7623,20,t") in new stack
[Apr 7 15:58:34] WARNING[14006]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'sip' (cause 20 - Unknown)
== Using SIP RTP CoS mark 5
[Apr 7 15:58:34] WARNING[14006]: acl.c:698 ast_ouraddrfor: Cannot connect
[Apr 7 15:58:34] WARNING[14006]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
-- Called 7624
-- Called 7623
[Apr 7 15:58:34] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 7 15:58:35] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 7 15:58:37] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 7 15:58:38] NOTICE[13914]: chan_iax2.c:4643 __auto_congest:
Auto-congesting call due to slow response
-- IAX2/0.0.29.199:4569-13525 is circuit-busy
-- Hungup 'IAX2/0.0.29.199:4569-13525'
[Apr 7 15:58:39] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x7fd08e3f45a0 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 7 15:58:40] WARNING[13920]: chan_sip.c:3386 retrans_pkt: Retransmission
timeout reached on transmission
6cf13d63561e7c106c31ffb74571e661@172.30.1.47:5060 for seqno 102 (Critical
Request) -- See doc/sip-retransmit.txt.
Packet timed out after 32000ms with no response
[Apr 7 15:58:41] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7527-0032' in macro 'stdexten'
== Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0032'
[Apr 7 15:58:49] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2e08d50 (len 789) to 0.0.29.200:5060 returned -1: Invalid argument
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 4 Apr 2011 20:22:55 +
Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Thanks for reply!
I found this problem only with X-lite version of softphone. Other phones are
working fine without any WARNING! look like X-lite has some short of SIP
issue.
-S
> From: mden...@gmail.com
> Date: Mon, 4 Apr 2011 15:59:43 -0400
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
>
> On Mon, Apr 4, 2011 at 3:51 PM, satish patel wrote:
> >
> > Hey Guys,
> >
> > Whenever i calling any extension i am getting following WARNING messages do
> > you have any idea they coming from where?
> >
> > -Satish
> >
> >
> >
> > shirley*CLI>
> > == Using SIP RTP CoS mark 5
> > -- Executing [7623@from-sip:1] Macro("SIP/7527-0008",
> > "stdexten,7623,sip/7623&sip/7624") in new stack
> > -- Executing [s@macro-stdexten:1] Dial("SIP/7527-0008",
> > "sip/7623&sip/7624&iax2/7623,20,t") in new stack
> > == Using SIP RTP CoS mark 5
> > -- Called 7623
> > == Using SIP RTP CoS mark 5
> > [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect
> > [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > -- Called 7624
> > -- Called 7623
> > -- SIP/7623-0009 is ringing
> > [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest:
> > Auto-congesting call due to slow response
> > -- IAX2/0.0.29.199:4569-5537 is circuit-busy
> > -- Hungup 'IAX2/0.0.29.199:4569-5537'
> > [Apr 4 12:46:45]
Re: [asterisk-users] Asterisk 1.8.3
Right now I'm testing 1.8.3 in devlopment and respose it pretty good
without realtime. (I didn't set realtime).
I ran stress test with sipp and pass 5000 call with RTP and no issue
at all. I got hogging at system resource but no issue at asterisk.
Look like I might go with 1.8.3 and later upgrade with 1.8.4 asap.
--
Sent from my iPhone
On Apr 7, 2011, at 9:12 AM, "Bryant Zimmerman"
wrote:
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik wrote:
> On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
>>
>> On Apr 6, 2011, at 8:54 PM, Edwin Lam
>> wrote:
>>
>>> On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
>>>>
>>>> Thanks for your response. I have added the patch for 18818 per
>>>> Michel Verbrask's
>>>> recomendation. It appers that it has made quite a difference. I
>>>> don't have an PRI
>>>> connections as all of our PRI's are connected via SIP gateways. I
>>>> did run into
>>>> serveral instances wher I had to kill -9 the process as well but
>>>> post patch I have
>>>> been in good shape know on wood. I hope there will be a new
>> release
>>>> that will
>>>> address the stability issues very soon if they release 1.8.4
>>>> without cleaning this
>>>> up I won't move unitl it is addressed.
>>>
>>> looking back at the messages file for the past 2 days. it
>>> just hanged on totally different events none of which related
>>> to Local channels.
>>>
>>> as far as the PRI not hearing early media issue. here's the
>>> excerpt from the messages file after "pri debug on" command:
>>>
>>> *
>>>
>>> -- Executing [18008291011@out_going_x:1] Dial("SIP/
>>
>> ... Parts Removed see origional response
>>
>>> -- Processing IE 30 (cs0, Progress Indicator)
>>> -- PROGRESS with cause code 127 received
>>> -- DAHDI/34-1 is making progress passing it to SIP/4988-6-0b45
>>>
>>> ***
>>>
>>> i used the same SIP station to dial the same 800 number
>>> on both versions (1.8.3.2 & 1.6.2.17). the output are
>>> pretty much identical except on 1.8.3.2, after the
>>> "PROGRESS with cause code 127..." message. i would hear
>>> nothing until the other side timed out & hang up, whereas on
>>> 1.6.2.17. i got the "DAHDI/... is making progress passing it to
>>> SIP..."
>>> message and can hear the early media from the other side.
>>>
>>>
>>>> For Now 1.8.3..2 is very bad.
>>>
>>> agreed...
>>
>> From: "Satish Patel"
>> Sent: Thursday, April 07, 2011 8:22 AM
>> Oh! Boy,
>>
>> Is it ture 1.8.3 is unstable? We are planning to put this in
>> production. Please suggest me what should I do?
>>
>>
>> Satish
>>
>> For me 1.8.3.2 has been the worst build that I have tried to use as
>> far a stability in a very long time. We are having issues
>> with deadlocks and voicemail.
>> I don't have a good option for you if you want to run 1.8 currently
>> the most stable release version I have found is 1.8.2.3 but I am
>> having the Voicemail issues there as well.
>> Things like messages not deleting propperly and hanging up the mail
>> box so users can't check them.
>
> 1.8.2 is unusable if you use RealTime without the patch in this
issue
>
> https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
>
>
From: "Satish Patel"
Sent: Thursday, April 07, 2011 9:06 AM
We don't have realtime configuration everything is in plain text file.
Is 1.8.3 has realtime issue or general issue?
Satish
I have seen my issues with the realtime disabled and using just
plain text. The issues get worse for me when we move to our realtime
confgs. So from my perspective I would say you might get farther
with realtime off but I would not bank on it.
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Re: [asterisk-users] Asterisk 1.8.3
We don't have realtime configuration everything is in plain text file.
Is 1.8.3 has realtime issue or general issue?
--
Sent from my iPhone
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik wrote:
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18818 per
Michel Verbrask's
recomendation. It appers that it has made quite a difference. I
don't have an PRI
connections as all of our PRI's are connected via SIP gateways. I
did run into
serveral instances wher I had to kill -9 the process as well but
post patch I have
been in good shape know on wood. I hope there will be a new
release
that will
address the stability issues very soon if they release 1.8.4
without cleaning this
up I won't move unitl it is addressed.
looking back at the messages file for the past 2 days. it
just hanged on totally different events none of which related
to Local channels.
as far as the PRI not hearing early media issue. here's the
excerpt from the messages file after "pri debug on" command:
*
-- Executing [18008291011@out_going_x:1] Dial("SIP/
... Parts Removed see origional response
-- Processing IE 30 (cs0, Progress Indicator)
-- PROGRESS with cause code 127 received
-- DAHDI/34-1 is making progress passing it to SIP/4988-6-0b45
***
i used the same SIP station to dial the same 800 number
on both versions (1.8.3.2 & 1.6.2.17). the output are
pretty much identical except on 1.8.3.2, after the
"PROGRESS with cause code 127..." message. i would hear
nothing until the other side timed out & hang up, whereas on
1.6.2.17. i got the "DAHDI/... is making progress passing it to
SIP..."
message and can hear the early media from the other side.
For Now 1.8.3..2 is very bad.
agreed...
From: "Satish Patel"
Sent: Thursday, April 07, 2011 8:22 AM
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put this in
production. Please suggest me what should I do?
Satish
For me 1.8.3.2 has been the worst build that I have tried to use as
far a stability in a very long time. We are having issues
with deadlocks and voicemail.
I don't have a good option for you if you want to run 1.8 currently
the most stable release version I have found is 1.8.2.3 but I am
having the Voicemail issues there as well.
Things like messages not deleting propperly and hanging up the mail
box so users can't check them.
1.8.2 is unusable if you use RealTime without the patch in this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Asterisk 1.8.3
Holy cow!!
Can I just build 1.8.2 over existing 1.8.3 ?
When we are going to fix all this thing???
--
Sent from my iPhone
On Apr 7, 2011, at 8:37 AM, "Bryant Zimmerman"
wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam
wrote:
> On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
>>
>> Thanks for your response. I have added the patch for 18818 per
>> Michel Verbrask's
>> recomendation. It appers that it has made quite a difference. I
>> don't have an PRI
>> connections as all of our PRI's are connected via SIP gateways. I
>> did run into
>> serveral instances wher I had to kill -9 the process as well but
>> post patch I have
>> been in good shape know on wood. I hope there will be a new release
>> that will
>> address the stability issues very soon if they release 1.8.4
>> without cleaning this
>> up I won't move unitl it is addressed.
>
> looking back at the messages file for the past 2 days. it
> just hanged on totally different events none of which related
> to Local channels.
>
> as far as the PRI not hearing early media issue. here's the
> excerpt from the messages file after "pri debug on" command:
>
> *
>
> -- Executing [18008291011@out_going_x:1] Dial("SIP/
... Parts Removed see origional response
> -- Processing IE 30 (cs0, Progress Indicator)
> -- PROGRESS with cause code 127 received
> -- DAHDI/34-1 is making progress passing it to SIP/4988-6-0b45
>
> ***
>
> i used the same SIP station to dial the same 800 number
> on both versions (1.8.3.2 & 1.6.2.17). the output are
> pretty much identical except on 1.8.3.2, after the
> "PROGRESS with cause code 127..." message. i would hear
> nothing until the other side timed out & hang up, whereas on
> 1.6.2.17. i got the "DAHDI/... is making progress passing it to
> SIP..."
> message and can hear the early media from the other side.
>
>
>> For Now 1.8.3..2 is very bad.
>
> agreed...
From: "Satish Patel"
Sent: Thursday, April 07, 2011 8:22 AM
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put this in
production. Please suggest me what should I do?
Satish
For me 1.8.3.2 has been the worst build that I have tried to use as
far a stability in a very long time. We are having issues with
deadlocks and voicemail.
I don't have a good option for you if you want to run 1.8 currently
the most stable release version I have found is 1.8.2.3 but I am
having the Voicemail issues there as well.
Things like messages not deleting propperly and hanging up the mail
box so users can't check them.
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Re: [asterisk-users] Asterisk 1.8.3
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put this in
production. Please suggest me what should I do?
--
Sent from my iPhone
On Apr 6, 2011, at 8:54 PM, Edwin Lam
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18818 per
Michel Verbrask's
recomendation. It appers that it has made quite a difference. I
don't have an PRI
connections as all of our PRI's are connected via SIP gateways. I
did run into
serveral instances wher I had to kill -9 the process as well but
post patch I have
been in good shape know on wood. I hope there will be a new release
that will
address the stability issues very soon if they release 1.8.4
without cleaning this
up I won't move unitl it is addressed.
looking back at the messages file for the past 2 days. it
just hanged on totally different events none of which related
to Local channels.
as far as the PRI not hearing early media issue. here's the
excerpt from the messages file after "pri debug on" command:
*
-- Executing [18008291011@out_going_x:1] Dial("SIP/
4988-6-0b45", "DAHDI/r1/18008291011,,f") in new stack
-- Making new call for cref 32974
-- Requested transfer capability: 0x00 - SPEECH
> DL-DATA request
> Protocol Discriminator: Q.931 (8) len=51
> TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent from originator)
> Message Type: SETUP (5)
TEI=0 Transmitting N(S)=87, window is open V(A)=87 K=7
> Protocol Discriminator: Q.931 (8) len=51
> TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent from originator)
> Message Type: SETUP (5)
> [04 03 80 90 a2]
> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
> Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
>User information layer 1: u-Law (34)
> [18 03 a1 83 8a]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare:
0 Preferred Dchan: 0
> ChanSel: As indicated in following octets
> Ext: 1 Coding: 0 Number Specified Channel
Type: 3
> Ext: 1 Channel: 10 Type: CPE]
> [28 06 b1 45 64 77 69 6e]
> Display (len= 6) Charset: 31 [ Edwin ]
> [6c 0c 21 83 34 31 35 34 33 39 34 39 38 38]
> Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
> Presentation: Presentation allowed of
network provided number (3) '4154394988' ]
> [70 0c 80 31 38 30 30 38 32 39 31 30 31 31]
> Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0)
NPI: Unknown Number Plan (0) '18008291011' ]
q931.c:5039 q931_setup: Call 32974 enters state 1 (Call Initiated).
Hold state: Idle
-- Called r1/18008291011
< Protocol Discriminator: Q.931 (8) len=13
< TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent to originator)
< Message Type: STATUS (125)
< [08 03 80 ab 28]
< Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare:
0 Location: User (0)
< Ext: 1 Cause: Access information discarded (43),
class = Network Congestion (resource unavailable) (2) ]
< Cause data 1: 28 (40)
< [14 01 01]
< Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0)
Call state: Call Initiated (1)
Received message for call 0x8fd2298 on 0x90d9cf0 TEI/SAPI 0/0, call-
>pri is 0x90d9cf0 TEI/SAPI 0/0
-- Processing IE 8 (cs0, Cause)
-- Processing IE 20 (cs0, Call State)
< Protocol Discriminator: Q.931 (8) len=10
< TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent to originator)
< Message Type: CALL PROCEEDING (2)
< [18 03 a9 83 8a]
< Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare:
0 Exclusive Dchan: 0
< ChanSel: As indicated in following octets
< Ext: 1 Coding: 0 Number Specified Channel
Type: 3
< Ext: 1 Channel: 10 Type: CPE]
Received message for call 0x8fd2298 on 0x90d9cf0 TEI/SAPI 0/0, call-
>pri is 0x90d9cf0 TEI/SAPI 0/0
-- Processing IE 24 (cs0, Channel Identification)
q931.c:7104 post_handle_q931_message: Call 32974 enters state 3
(Outgoing Call Proceeding). Hold state: Idle
-- DAHDI/34-1 is proceeding passing it to SIP/4988-6-0b45
< Protocol Discriminator: Q.931 (8) len=13
< TEI=0 Call Ref: len= 2 (reference 206/0xCE) (Sent to originator)
< Message Type: PROGRESS (3)
< [08 02 82 ff]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare:
0 Location: Public network serving the local user (2)
< Ext: 1 Cause: Interworking, unspecified (127),
class = Interworking (7) ]
< [1e 02 82 81]
< Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0 Location: Public network serving the local user (2)
< Ext: 1 Progress Description: Call
is not end-to-end ISDN; further call progress information may be
available inband. (1
Re: [asterisk-users] asterisk meetme invalid extension
i did and its not working here is console output. We have 8910-8920 meetme
conf room. below i am dialing 8991 for test invalid and its not working..
Packet timed out after 32000ms with no response
== Using SIP RTP CoS mark 5
-- Executing [7580@from-sip:1] Goto("SIP/7527-0030", "ivr-meetme,s,1")
in new stack
-- Goto (ivr-meetme,s,1)
-- Executing [s@ivr-meetme:1] Answer("SIP/7527-0030", "") in new stack
-- Executing [s@ivr-meetme:2] Wait("SIP/7527-0030", "1") in new stack
-- Executing [s@ivr-meetme:3] BackGround("SIP/7527-0030",
"conf-getconfno") in new stack
-- Playing 'conf-getconfno.ulaw' (language 'en')
-- Executing [s@ivr-meetme:4] WaitExten("SIP/7527-0030", "20") in new
stack
== CDR updated on SIP/7527-0030
-- Executing [8991@ivr-meetme:1] MeetMe("SIP/7527-0030", "8991,cMp") in
new stack
== Parsing '/etc/asterisk/meetme.conf': == Found
== Spawn extension (ivr-meetme, 8991, 1) exited non-zero on
'SIP/7527-0030'
shirley*CLI>
> Date: Wed, 6 Apr 2011 14:37:20 -0700
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk meetme invalid extension
>
> On Wed, 6 Apr 2011, satish patel wrote:
>
> > I have following dialplan for meetme and i want if someone type wrong
> > meetme extension it should say invalid extension. But look like
> > following doesn't work. its just hangup if i type wrong number. how to
> > fix this code..
> >
> > exten => i,n,Playback(pbx-invalid)
>
> The priority should be 1.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
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>
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[asterisk-users] asterisk meetme invalid extension
Hey Guy!
I have following dialplan for meetme and i want if someone type wrong meetme
extension it should say invalid extension. But look like following doesn't
work. its just hangup if i type wrong number. how to fix this code..
;Conference rooms/lines:
exten => 7580,1,Goto(ivr-meetme,s,1)
[ivr-meetme]
include => meetme
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,Background(conf-getconfno)
exten => s,n,WaitExten(20)
exten => s,n,Hangup()
exten => i,n,Playback(pbx-invalid)
[meetme]
exten => _89XX,1,MeetMe(${EXTEN},cMp)
--
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Re: [asterisk-users] [SOLVED] IAX trunk error AES encryption disabled. Install OpenSSL.
res_crypto module was not loaded :)
Whenever i post question and after few min i got answer myself. Magic Sorry
for bother you..
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 6 Apr 2011 19:59:02 +
Subject: Re: [asterisk-users] IAS trunk error AES encryption disabled. Install
OpenSSL.
also i have linssl-dev
root@shirley:/usr/local/src/asterisk/asterisk-1.8.3.2/contrib/scripts# dpkg -l
| grep ssl
ii libssl-dev 0.9.8k-7ubuntu8.6 SSL
development libraries, header files and
ii libssl0.9.8 0.9.8k-7ubuntu8.6 SSL
shared libraries
ii openssl 0.9.8k-7ubuntu8.6 Secure
Socket Layer (SSL) binary and related
ii python-openssl 0.10-1Python
wrapper around the OpenSSL library
ii ssl-cert1.0.23ubuntu2 simple
debconf wrapper for OpenSSL
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 6 Apr 2011 19:53:41 +
Subject: Re: [asterisk-users] IAS trunk error AES encryption disabled. Install
OpenSSL.
Yes, I do have that install.
root@shirley:/var/lib/asterisk/agi-bin# dpkg -l | grep -i openssl
ii openssl 0.9.8k-7ubuntu8.6 Secure
Socket Layer (SSL) binary and related
ii python-openssl 0.10-1Python
wrapper around the OpenSSL library
ii ssl-cert1.0.23ubuntu2 simple
debconf wrapper for OpenSSL
Date: Wed, 6 Apr 2011 14:48:55 -0500
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAS trunk error AES encryption disabled. Install
OpenSSL.
On Wed, Apr 6, 2011 at 2:45 PM, satish patel wrote:
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-000d",
"orasebcamdial,7623") in new stack
-- Executing [s@macro-orasebcamdial:1] Dial("SIP/7527-000d",
"iax2/orasebcam@orasebcam/7623") in new stack
-- Called orasebcam@orasebcam/7623
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:145
__stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL.
Do you have OpenSSL installed?
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--
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Re: [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.
also i have linssl-dev
root@shirley:/usr/local/src/asterisk/asterisk-1.8.3.2/contrib/scripts# dpkg -l
| grep ssl
ii libssl-dev 0.9.8k-7ubuntu8.6 SSL
development libraries, header files and
ii libssl0.9.8 0.9.8k-7ubuntu8.6 SSL
shared libraries
ii openssl 0.9.8k-7ubuntu8.6 Secure
Socket Layer (SSL) binary and related
ii python-openssl 0.10-1Python
wrapper around the OpenSSL library
ii ssl-cert1.0.23ubuntu2 simple
debconf wrapper for OpenSSL
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 6 Apr 2011 19:53:41 +
Subject: Re: [asterisk-users] IAS trunk error AES encryption disabled. Install
OpenSSL.
Yes, I do have that install.
root@shirley:/var/lib/asterisk/agi-bin# dpkg -l | grep -i openssl
ii openssl 0.9.8k-7ubuntu8.6 Secure
Socket Layer (SSL) binary and related
ii python-openssl 0.10-1Python
wrapper around the OpenSSL library
ii ssl-cert1.0.23ubuntu2 simple
debconf wrapper for OpenSSL
Date: Wed, 6 Apr 2011 14:48:55 -0500
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAS trunk error AES encryption disabled. Install
OpenSSL.
On Wed, Apr 6, 2011 at 2:45 PM, satish patel wrote:
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-000d",
"orasebcamdial,7623") in new stack
-- Executing [s@macro-orasebcamdial:1] Dial("SIP/7527-000d",
"iax2/orasebcam@orasebcam/7623") in new stack
-- Called orasebcam@orasebcam/7623
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:145
__stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL.
Do you have OpenSSL installed?
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--
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Re: [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.
Yes, I do have that install.
root@shirley:/var/lib/asterisk/agi-bin# dpkg -l | grep -i openssl
ii openssl 0.9.8k-7ubuntu8.6 Secure
Socket Layer (SSL) binary and related
ii python-openssl 0.10-1Python
wrapper around the OpenSSL library
ii ssl-cert1.0.23ubuntu2 simple
debconf wrapper for OpenSSL
Date: Wed, 6 Apr 2011 14:48:55 -0500
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAS trunk error AES encryption disabled. Install
OpenSSL.
On Wed, Apr 6, 2011 at 2:45 PM, satish patel wrote:
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-000d",
"orasebcamdial,7623") in new stack
-- Executing [s@macro-orasebcamdial:1] Dial("SIP/7527-000d",
"iax2/orasebcam@orasebcam/7623") in new stack
-- Called orasebcam@orasebcam/7623
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:145
__stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL.
Do you have OpenSSL installed?
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--
_
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Re: [asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.
Look like this issue is still there.
From: satish...@hotmail.com
To: satish...@hotmail.com
Subject: RE: IAS trunk error AES encryption disabled. Install OpenSSL.
Date: Wed, 6 Apr 2011 19:45:06 +
look like this issue is still there
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: IAS trunk error AES encryption disabled. Install OpenSSL.
Date: Tue, 5 Apr 2011 20:54:43 +
Hey Guys!
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-000d",
"orasebcamdial,7623") in new stack
-- Executing [s@macro-orasebcamdial:1] Dial("SIP/7527-000d",
"iax2/orasebcam@orasebcam/7623") in new stack
-- Called orasebcam@orasebcam/7623
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:145
__stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL.
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:157
__stub__ast_aes_set_decrypt_key: AES encryption disabled. Install OpenSSL.
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:157
__stub__ast_aes_set_decrypt_key: AES encryption disabled. Install OpenSSL.
-- Call accepted by 172.30.245.208 (format gsm)
-- Format for call is gsm
-- IAX2/orasebcam-16782 is ringing
-- IAX2/orasebcam-16782 is circuit-busy
-- Hungup 'IAX2/orasebcam-16782'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-orasebcamdial:2] Goto("SIP/7527-000d",
"s-CONGESTION,1") in new stack
--
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Re: [asterisk-users] asterisk hints
You are right i believe, My Polycom 501 not sending subscription to asterisk. shirley*CLI> sip show subscriptions Peer User Call ID ExtensionLast state TypeMailboxExpiry 0 active SIP subscriptions shirley*CLI> Date: Wed, 6 Apr 2011 18:48:55 +0200 From: oza_4...@yahoo.fr To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk hints 2011/4/5 Danny Nicholas On my Polycom 501’s I use hints to populate a “buddy” list – I hit the buddies softkey and can see if my “buddy” is on the line. Hi, Sorry to hijack this thread but are your Ringing phones displayed as InUse ones with your setup ? My understanding of this is : 1. Polycom 3.2 firmware brings the capability to have a third state (beside InUse and Idle) but this firmware is not available for 501's. 2. It is possible to get Ringing status with Polycom 3.1 firmware but you need a kind of Notify/Subscribe which is not yet implemented in Asterisk. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk hints
But i need to see all my extension state: Idle or Inuse
How should i monitor all my phone with hint catch-all _XXX
If you have example please post me.
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Wed, 6 Apr 2011 10:12:04 -0500
Subject: Re: [asterisk-users] asterisk hints
This will only generate hints for
7400-7699.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Wednesday, April 06, 2011
10:09 AM
To: asterisk-users
Subject: Re: [asterisk-users]
asterisk hints
I used following hint dialplan and i ran show hints but its
showing only one extension what about other 200 phones status ?
exten => _7[456]XX,hint,SIP/${EXTEN}
exten => _7[456]XX,1,Macro(stdexten,${EXTEN},sip/${EXTEN})
shirley*CLI> core show hints
-= Registered Asterisk Dial Plan Hints =-
_7[456]XX@ora-cam-extensions :
SIP/${EXTEN}
State:Idle
Watchers 0
- 1 hints registered
> Date: Wed, 6 Apr 2011 15:25:08 +0200
> From: s...@sil.at
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk hints
>
> Am 05.04.11 20:35, schrieb satish patel:
> >
> > If i want to watch every phone status Idel or Inuse the how should i
use hint in my dialplan. I meant should i need to specify each and every
extension ? or is there any catch-all extensions ?
> >
> > -Satish
> >
> Hello,
>
> You can use a hint wildcard like _XXX the _ is important cause this
> means that this hint is a dynamic hint.
>
> for every subscribe which match the dynamic hint you will see a normal
> hint which is created by asterisk itself.
>
> best regards
>
> Stefan
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] asterisk hints
I used following hint dialplan and i ran show hints but its showing only one
extension what about other 200 phones status ?
exten => _7[456]XX,hint,SIP/${EXTEN}
exten => _7[456]XX,1,Macro(stdexten,${EXTEN},sip/${EXTEN})
shirley*CLI> core show hints
-= Registered Asterisk Dial Plan Hints =-
_7[456]XX@ora-cam-extensions : SIP/${EXTEN} State:Idle
Watchers 0
- 1 hints registered
> Date: Wed, 6 Apr 2011 15:25:08 +0200
> From: s...@sil.at
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk hints
>
> Am 05.04.11 20:35, schrieb satish patel:
> >
> > If i want to watch every phone status Idel or Inuse the how should i use
> > hint in my dialplan. I meant should i need to specify each and every
> > extension ? or is there any catch-all extensions ?
> >
> > -Satish
> >
> Hello,
>
> You can use a hint wildcard like _XXX the _ is important cause this
> means that this hint is a dynamic hint.
>
> for every subscribe which match the dynamic hint you will see a normal
> hint which is created by asterisk itself.
>
> best regards
>
> Stefan
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] IAS trunk error AES encryption disabled. Install OpenSSL.
Hey Guys!
I am getting this wired error when i am calling IAX trunk. Everything works!
but i want to get rid on these RED WARNING messages.. what is wrong here ? I
have func_aes.so module loaded. also i remove and test but still same error.
-Satish
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-000d",
"orasebcamdial,7623") in new stack
-- Executing [s@macro-orasebcamdial:1] Dial("SIP/7527-000d",
"iax2/orasebcam@orasebcam/7623") in new stack
-- Called orasebcam@orasebcam/7623
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:145
__stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL.
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:157
__stub__ast_aes_set_decrypt_key: AES encryption disabled. Install OpenSSL.
[Apr 5 13:51:26] WARNING[9539]:
/usr/local/src/asterisk/asterisk-1.8.3.2/include/asterisk/crypto.h:157
__stub__ast_aes_set_decrypt_key: AES encryption disabled. Install OpenSSL.
-- Call accepted by 172.30.245.208 (format gsm)
-- Format for call is gsm
-- IAX2/orasebcam-16782 is ringing
-- IAX2/orasebcam-16782 is circuit-busy
-- Hungup 'IAX2/orasebcam-16782'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-orasebcamdial:2] Goto("SIP/7527-000d",
"s-CONGESTION,1") in new stack
--
_
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Re: [asterisk-users] asterisk hints
If i want to watch every phone status Idel or Inuse the how should i use hint
in my dialplan. I meant should i need to specify each and every extension ? or
is there any catch-all extensions ?
-Satish
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 13:20:45 -0500
Subject: Re: [asterisk-users] asterisk hints
On my Polycom 501’s I use hints to
populate a “buddy” list – I hit the buddies softkey and can
see if my “buddy” is on the line.
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011 1:19
PM
To: asterisk-users
Subject: Re: [asterisk-users]
asterisk hints
I am using asterisk-1.8.3.2
and we have polycom phones. how should i use hint ?
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 12:56:58 -0500
Subject: Re: [asterisk-users] asterisk hints
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011
12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk
hints
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should
i type each and every extension in hint dialplan or is there regex available
something like following _XXX will watch all my extension. Because we
have more than 200 phones so its hard to write down each and every extension in
hint
[hints]
exten => _XXX,hint,SIP/${EXTEN}
exten => 7527,hint,SIP/7527
The
answer depends on the version you are using. Hints are (in my experience)
most useful for BLF and AMI applications.
-- _ --
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Re: [asterisk-users] asterisk hints
I am using asterisk-1.8.3.2
and we have polycom phones. how should i use hint ?
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 12:56:58 -0500
Subject: Re: [asterisk-users] asterisk hints
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Tuesday, April 05, 2011
12:54 PM
To: asterisk-users
Subject: [asterisk-users] asterisk
hints
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should i type each and every extension in hint dialplan or is there regex
available
something like following _XXX will watch all my extension. Because we
have more than 200 phones so its hard to write down each and every extension in
hint
[hints]
exten => _XXX,hint,SIP/${EXTEN}
exten => 7527,hint,SIP/7527
The
answer depends on the version you are using. Hints are (in my experience) most
useful for BLF and AMI applications.
--
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[asterisk-users] asterisk hints
Hey guys!
I am new in hints application. what is the use of this application ( i already
did google ) but still confused. If i want to use hint in my dialplan then
should i type each and every extension in hint dialplan or is there regex
available
something like following _XXX will watch all my extension. Because we have
more than 200 phones so its hard to write down each and every extension in hint
[hints]
exten => _XXX,hint,SIP/${EXTEN}
exten => 7527,hint,SIP/7527
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Re: [asterisk-users] allpage issu on asterisk 1.8.3.x
Nevermind,
I have solved it my self. this script wring some logs in /tmp and somehow
logfile was already there. so just deleted and it works!
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 5 Apr 2011 16:35:37 +
Subject: [asterisk-users] allpage issu on asterisk 1.8.3.x
Hey Guys!
I have perl script for allpage which is working fine with asterisk 1.8.2.3
version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is
there anything changes ?
If i run this script from command like it works but not from asterisk dialplan.
This script nothing but just connecting AMI interface and using Variable:
SIPADDHEADER=Alert-Info: Ring Answer variable to call all phones and putting
them in meetme conf room.
following is sample of script ( I am pasting half script )
# Now, we have an array (@tocall) with all valid SIP extensions.
while (my $sipxtn = shift @tocall) {
print "VERBOSE \"Doing $sipxtn\" 0\n";
# Open connection to AGI
my $tn = new Net::Telnet ( Port => $mgrport,
Prompt => '/.*[\$%#>] $/',
Output_record_separator => '',
Input_Log=> "/tmp/input.log",
Output_Log=> "/tmp/output.log",
Errmode=> 'return', );
$tn->open("127.0.0.1");
$tn->waitfor('/0\n$/');
$tn->print("Action: Login\n");
$tn->print("Username: $mgruser\n");
$tn->print("Secret: $mgrpass\n");
$tn->print("Events: off\n\n");
my ($pm, $m) = $tn->waitfor('/Authentication (.+)\n\n/');
if ($m =~ /Authentication failed/) {
print "VERBOSE \"Incorrect MGRUSER or MGRPASS - unable to connect to
manager interface\" 0\n";
exit;
}
$tn->print("Action: Originate\nChannel: SIP/$sipxtn\nContext:
all-page\nPriority: 1\n");
$tn->print("Variable: SIPADDHEADER=Alert-Info: Ring Answer\n");
$tn->print("Extension: s\n");
$tn->print("CallerID: System Page\n");
$tn->print("Action: Logoff\n\n");
$tn->close;
}
-S
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[asterisk-users] allpage issu on asterisk 1.8.3.x
Hey Guys!
I have perl script for allpage which is working fine with asterisk 1.8.2.3
version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is
there anything changes ?
If i run this script from command like it works but not from asterisk dialplan.
This script nothing but just connecting AMI interface and using Variable:
SIPADDHEADER=Alert-Info: Ring Answer variable to call all phones and putting
them in meetme conf room.
following is sample of script ( I am pasting half script )
# Now, we have an array (@tocall) with all valid SIP extensions.
while (my $sipxtn = shift @tocall) {
print "VERBOSE \"Doing $sipxtn\" 0\n";
# Open connection to AGI
my $tn = new Net::Telnet ( Port => $mgrport,
Prompt => '/.*[\$%#>] $/',
Output_record_separator => '',
Input_Log=> "/tmp/input.log",
Output_Log=> "/tmp/output.log",
Errmode=> 'return', );
$tn->open("127.0.0.1");
$tn->waitfor('/0\n$/');
$tn->print("Action: Login\n");
$tn->print("Username: $mgruser\n");
$tn->print("Secret: $mgrpass\n");
$tn->print("Events: off\n\n");
my ($pm, $m) = $tn->waitfor('/Authentication (.+)\n\n/');
if ($m =~ /Authentication failed/) {
print "VERBOSE \"Incorrect MGRUSER or MGRPASS - unable to connect to
manager interface\" 0\n";
exit;
}
$tn->print("Action: Originate\nChannel: SIP/$sipxtn\nContext:
all-page\nPriority: 1\n");
$tn->print("Variable: SIPADDHEADER=Alert-Info: Ring Answer\n");
$tn->print("Extension: s\n");
$tn->print("CallerID: System Page\n");
$tn->print("Action: Logoff\n\n");
$tn->close;
}
-S
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Re: [asterisk-users] Read VoiceMail direct
Perfect! Thanks
what about :-4 ? I want to remove some digits
-satish
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 4 Apr 2011 23:16:30 +0200
Subject: Re: [asterisk-users] Read VoiceMail direct
Hi,
maybe:
exten =>
8500,3,VoiceMailMain(${CALLERID(num)}@default)
Regards
- Andrea
- Original Message -
From:
satish
patel
To: asterisk-users
Sent: Monday, April 04, 2011 11:08
PM
Subject: [asterisk-users] Read VoiceMail
direct
Hey Guy!
I want direct access of VoiceMail without
asking mailbox number (Direct ask PIN). I am using following dialplan but its
still asking me Mailbox number. Look like asterisk 1.8 doesn't support
CALLERIDNUM variable.
Do you have any idea ?
exten =>
8500,1,answer
exten => 8500,2,wait(1)
exten =>
8500,3,voicemailmain(${CALLERIDNUM:-4}@default)
exten =>
8500,4,hangup
exten => i,1,playback(invalid)
exten =>
i,2,hangup
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[asterisk-users] Read VoiceMail direct
Hey Guy!
I want direct access of VoiceMail without asking mailbox number (Direct ask
PIN). I am using following dialplan but its still asking me Mailbox number.
Look like asterisk 1.8 doesn't support CALLERIDNUM variable.
Do you have any idea ?
exten => 8500,1,answer
exten => 8500,2,wait(1)
exten => 8500,3,voicemailmain(${CALLERIDNUM:-4}@default)
exten => 8500,4,hangup
exten => i,1,playback(invalid)
exten => i,2,hangup
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Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Thanks for reply!
I found this problem only with X-lite version of softphone. Other phones are
working fine without any WARNING! look like X-lite has some short of SIP
issue.
-S
> From: mden...@gmail.com
> Date: Mon, 4 Apr 2011 15:59:43 -0400
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
>
> On Mon, Apr 4, 2011 at 3:51 PM, satish patel wrote:
> >
> > Hey Guys,
> >
> > Whenever i calling any extension i am getting following WARNING messages do
> > you have any idea they coming from where?
> >
> > -Satish
> >
> >
> >
> > shirley*CLI>
> > == Using SIP RTP CoS mark 5
> > -- Executing [7623@from-sip:1] Macro("SIP/7527-0008",
> > "stdexten,7623,sip/7623&sip/7624") in new stack
> > -- Executing [s@macro-stdexten:1] Dial("SIP/7527-0008",
> > "sip/7623&sip/7624&iax2/7623,20,t") in new stack
> > == Using SIP RTP CoS mark 5
> > -- Called 7623
> > == Using SIP RTP CoS mark 5
> > [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect
> > [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > -- Called 7624
> > -- Called 7623
> > -- SIP/7623-0009 is ringing
> > [Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest:
> > Auto-congesting call due to slow response
> > -- IAX2/0.0.29.199:4569-5537 is circuit-busy
> > -- Hungup 'IAX2/0.0.29.199:4569-5537'
> > [Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > -- SIP/7623-0009 connected line has changed. Saving it until answer
> > for SIP/7527-0008
> > -- SIP/7623-0009 answered SIP/7527-0008
> > [Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > == Spawn extension (macro-stdexten, s, 1) exited non-zero on
> > 'SIP/7527-0008' in macro 'stdexten'
> > == Spawn extension (from-sip, 7623, 1) exited non-zero on
> > 'SIP/7527-0008'
> > [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
> > 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
> > [Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission
> > timeout reached on transmission
> > 23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical
> > Request) -- See doc/sip-retransmit.txt.
> > Packet timed out after 32000ms with no response
> >
> >
>
> Satish,
>
> Run dmesg and look for anything funny. This sounds very similar to
> when I had a netfilter nat "helper" not helping me at all.
>
> -M
>
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[asterisk-users] WARNING chan_sip.c:3115 __sip_xmit
Hey Guys,
Whenever i calling any extension i am getting following WARNING messages do you
have any idea they coming from where?
-Satish
shirley*CLI>
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro("SIP/7527-0008",
"stdexten,7623,sip/7623&sip/7624") in new stack
-- Executing [s@macro-stdexten:1] Dial("SIP/7527-0008",
"sip/7623&sip/7624&iax2/7623,20,t") in new stack
== Using SIP RTP CoS mark 5
-- Called 7623
== Using SIP RTP CoS mark 5
[Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect
[Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
-- Called 7624
-- Called 7623
-- SIP/7623-0009 is ringing
[Apr 4 12:46:38] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 4 12:46:39] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 4 12:46:41] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 4 12:46:41] NOTICE[5975]: chan_iax2.c:4643 __auto_congest:
Auto-congesting call due to slow response
-- IAX2/0.0.29.199:4569-5537 is circuit-busy
-- Hungup 'IAX2/0.0.29.199:4569-5537'
[Apr 4 12:46:45] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
-- SIP/7623-0009 connected line has changed. Saving it until answer for
SIP/7527-0008
-- SIP/7623-0009 answered SIP/7527-0008
[Apr 4 12:46:53] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7527-0008' in macro 'stdexten'
== Spawn extension (from-sip, 7623, 1) exited non-zero on 'SIP/7527-0008'
[Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument
[Apr 4 12:47:09] WARNING[5982]: chan_sip.c:3386 retrans_pkt: Retransmission
timeout reached on transmission
23bee79c00a393995398c4d76372049e@172.30.1.47:5060 for seqno 102 (Critical
Request) -- See doc/sip-retransmit.txt.
Packet timed out after 32000ms with no response
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Re: [asterisk-users] Polycom 501 alternate
You are awesome!!! -- Sent from my iPhone On Apr 1, 2011, at 5:40 PM, Warren Selby wrote: The Polycom 501 has basically been replaced by the Polycom 550. Thanks, --Warren Selby, dCAP On Apr 1, 2011, at 4:25 PM, satish patel wrote: We're looking to purchase new phones for Asterisk. There are a limited number of new Polycom 501's on the market, mostly refurbished available. Can you recommend a replacement phone? What ever model replaced the 501? -Satish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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
[asterisk-users] Polycom 501 alternate
We're looking to purchase new phones for Asterisk. There are a limited number of new Polycom 501's on the market, mostly refurbished available. Can you recommend a replacement phone? What ever model replaced the 501? -Satish -- _ -- 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] codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode
Ah! so Hardware Transcoder is separate hardware ?? This is not a PRI card right ? -Satish > Date: Fri, 1 Apr 2011 15:14:11 -0500 > From: sruff...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] codec_dahdi find_transcoders: Failed to open > /dev/dahdi/transcode > > On 04/01/2011 02:55 PM, satish patel wrote: > > > > I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am > > starting asterisk i am getting this error on console. > > > > > > func_callerid.so => (Party ID related dialplan functions (Caller-ID, > > Connected-line, Redirecting)) > > == Registered application 'PrivacyManager' > > app_privacy.so => (Require phone number to be entered, if no CallerID sent) > > == Registered custom function 'TIMEOUT' > > func_timeout.so => (Channel timeout dialplan functions) > > == Registered custom function 'CDR' > > func_cdr.so => (Call Detail Record (CDR) dialplan function) > > *[Apr 1 12:51:27] ERROR[21102]: codec_dahdi.c:578 find_transcoders: > > Failed to open /dev/dahdi/transcode: No such file or directory* > > You can either ignore that error message or add "noload => > codec_dahdi.so" to your modules.conf if you do not have a hardware > transcoder installed in your system. > > -- > Shaun Ruffell > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode
I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function 'TIMEOUT' func_timeout.so => (Channel timeout dialplan functions) == Registered custom function 'CDR' func_cdr.so => (Call Detail Record (CDR) dialplan function) [Apr 1 12:51:27] ERROR[21102]: codec_dahdi.c:578 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory codec_dahdi.so => (Generic DAHDI Transcoder Codec Translator) == Registered application 'While' == Registered application 'EndWhile' == Registered application 'ExitWhile' == Registered application 'ContinueWhile' -- _ -- 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] Hold problem with Queue
We need logs or console output
--
Sent from my iPhone
On Apr 1, 2011, at 9:01 AM, Elensarde wrote:
Yes, when the caller are in the queue
New informations :
- If A call B directly and B hold A, it's work...
- Test with Asterisk 1.8.0, 1.8.1, 1.8.2, same problems...
- Phones : Cisco SPA502G / SPA508G / SPA509G
2011/4/1 Satish Patel :
Do you have music on hold configure?
--
Sent from my iPhone
On Apr 1, 2011, at 3:39 AM, Elensarde wrote:
Hello List,
First, sorry for my bad English skill, I'm French.
We have an asterisk 1.8.3.2 built from sources with a simple Queue :
[TestQueue]
strategy=ringall
timeout=15
retry=1
timeoutpriority=conf
ringinuse=yes
wrapuptime=2
member => SIP/002E31,0,Agent A
member => SIP/1CA3F2,0,Agent B
member => SIP/E08972,0,Agent C
And this dialplan (extension.ael) :
3600 => {
Answer();
Queue(TestQueue60);
Playback(invalid);
Hangup();
}
When somebody call this exten, an Agent take the call without
problems.
But when he want hold this, phone try to hold the caller without
success.
Finally, no signal in the caller-line and the agent-line is hangup
(for the phone), I not have errors or warnings in logs...
Any ideas ?
Thanks in advance, and kind regards,
Elensarde
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Re: [asterisk-users] Best Scripting Language
Do you think C is a scripting language? -- Sent from my iPhone On Apr 1, 2011, at 8:27 AM, Roger Burton West wrote: On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote: Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Depends on the other parameters. Perl is great for rapid development, but I wouldn't run it per-call on a box taking hundreds of calls per second. (Ditto Ruby and Python.) C will be much faster, but it's more effort to write and debug. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Scripting Language
No doubt perl is best. But python getting more popular these days. -- Sent from my iPhone On Apr 1, 2011, at 8:00 AM, mahesh katta wrote: Perl is the best script On Fri, Apr 1, 2011 at 5:27 PM, Gopalakrishnan A.N wrote: Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- 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 -- Best Regards, Mahesh Katta BUZZWORKS Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 303, Gagangiri Apts, Parleshwar Road, Ville Parle East, Mumbai - 400057. GSM +91.97029.70779 | Phone +91.22.2663.1811 | Fax +91.22.2663.1811 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hold problem with Queue
Do you have music on hold configure?
--
Sent from my iPhone
On Apr 1, 2011, at 3:39 AM, Elensarde wrote:
Hello List,
First, sorry for my bad English skill, I'm French.
We have an asterisk 1.8.3.2 built from sources with a simple Queue :
[TestQueue]
strategy=ringall
timeout=15
retry=1
timeoutpriority=conf
ringinuse=yes
wrapuptime=2
member => SIP/002E31,0,Agent A
member => SIP/1CA3F2,0,Agent B
member => SIP/E08972,0,Agent C
And this dialplan (extension.ael) :
3600 => {
Answer();
Queue(TestQueue60);
Playback(invalid);
Hangup();
}
When somebody call this exten, an Agent take the call without
problems.
But when he want hold this, phone try to hold the caller without
success.
Finally, no signal in the caller-line and the agent-line is hangup
(for the phone), I not have errors or warnings in logs...
Any ideas ?
Thanks in advance, and kind regards,
Elensarde
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Re: [asterisk-users] chan_dahdi unknown dependency problem
Run pre requirement check script I don't know the name but it's located inside asterisk source dir inside contrib I had same issue and has been fixed by that. -- Sent from my iPhone On Mar 31, 2011, at 5:47 PM, "Kevin P. Fleming" wrote: On 03/30/2011 01:32 PM, SebA wrote: So, I've compiled and installed libpri-1.4.11.5, dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but chan_dahdi is not getting built. If I do a "make menuselect" in asterisk I see it listed with XXX, meaning that dependencies are not met. XXX chan_dahdi Depends on: res_smdi(M), dahdi(E), tonezone(E), pri(E), ss7(E), openr2(E) When I run 'make menuselect', this is what I see for chan_dahdi: DAHDI Telephony Depends on: res_smdi(M), dahdi(E), tonezone(E) Can use: pri(E), ss7(E), openr2(E) Yours says 'depends on' for all of these items, which means you *must* have them installed. Have you made any changes to the Asterisk source code? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s extension not working
@Sherwood,
I was also thinking about that But then how 's' extension match any unknown
number ? Like when call coming from PSTN then how IVR picked up...?
-Satish
> Date: Mon, 28 Mar 2011 12:58:28 -0500
> From: sherwood.mcgo...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] s extension not working
>
> Uhm
>
> That's because you're being passed 7527 as the extension, so it won't
> match "s"
>
> On 3/28/2011 11:38 AM, satish patel wrote:
> > If i use 's' then i got following error. This scenario is back to
> > back asterisk connected on PRI line (T1). for testing purpose i
> > calling from one asterisk to other and i want to land call on 's'
> > extension.
> >
> > shirley*CLI>
> > -- Extension '7527' in context 'from-pstn' from '7623' does not
> > exist. Rejecting call on channel 0/1, span 1
> >
> >
> >
> >
> > If i use _XXX then it working with following output.
> >
> > shirley*CLI>
> > -- Accepting call from '7623' to '7527' on channel 0/1, span 1
> > -- Executing [7527@from-pstn:1] Answer("DAHDI/i1/7623-10", "") in
> > new stack
> > -- Executing [7527@from-pstn:2] Playback("DAHDI/i1/7623-10",
> > "hello-world") in new stack
> > -- Playing 'hello-world.ulaw' (language 'en')
> > -- Executing [7527@from-pstn:3] Hangup("DAHDI/i1/7623-10", "") in
> > new stack
> > == Spawn extension (from-pstn, 7527, 3) exited non-zero on
> > 'DAHDI/i1/7623-10'
> > -- Hungup 'DAHDI/i1/7623-10'
> >
> >
> >
> > --------
> > From: da...@debsinc.com
> > To: asterisk-users@lists.digium.com
> > Date: Mon, 28 Mar 2011 11:08:57 -0500
> > Subject: Re: [asterisk-users] s extension not working
> >
> >
> >
> > *From:*asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish
> > patel
> > *Sent:* Monday, March 28, 2011 11:04 AM
> > *To:* asterisk-users
> > *Subject:* [asterisk-users] s extension not working
> >
> >
> >
> > Hey Guys!
> >
> > I have asterisk 1.8.x and somehow my 's' extension not picking up any
> > incoming calls..
> >
> > Not working
> >
> > [from-pstn]
> > exten => s,1,Answer()
> > same => n,Playback(hello-world)
> > same => n,Hangup()
> >
> >
> >
> >
> > Working...
> >
> > [from-pstn]
> > exten => _,1,Answer()
> > same => n,Playback(hello-world)
> > same => n,Hangup()
> >
> >
> > -S
> >
> >
> >
> > Ok Satish. I assume sip.conf or dahdi.conf has a context of
> > from-pstn. The key to actually solving this will be for you to give
> > us say 10 lines of CLI output.
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello asterisk-users mailing list To
> > UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> Sherwood McGowan
> Carrier, ITSP, Call Center, and PBX Solutions Consultant
>
>
> --
> _
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> asterisk-users mailing list
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Re: [asterisk-users] s extension not working
If i use 's' then i got following error. This scenario is back to back
asterisk connected on PRI line (T1). for testing purpose i calling from one
asterisk to other and i want to land call on 's' extension.
shirley*CLI>
-- Extension '7527' in context 'from-pstn' from '7623' does not exist.
Rejecting call on channel 0/1, span 1
If i use _XXX then it working with following output.
shirley*CLI>
-- Accepting call from '7623' to '7527' on channel 0/1, span 1
-- Executing [7527@from-pstn:1] Answer("DAHDI/i1/7623-10", "") in new stack
-- Executing [7527@from-pstn:2] Playback("DAHDI/i1/7623-10", "hello-world")
in new stack
-- Playing 'hello-world.ulaw' (language 'en')
-- Executing [7527@from-pstn:3] Hangup("DAHDI/i1/7623-10", "") in new stack
== Spawn extension (from-pstn, 7527, 3) exited non-zero on 'DAHDI/i1/7623-10'
-- Hungup 'DAHDI/i1/7623-10'
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 28 Mar 2011 11:08:57 -0500
Subject: Re: [asterisk-users] s extension not working
From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, March 28, 2011 11:04
AM
To: asterisk-users
Subject: [asterisk-users] s
extension not working
Hey Guys!
I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming
calls..
Not working
[from-pstn]
exten => s,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
Working...
[from-pstn]
exten => _,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
-S
Ok Satish. I assume sip.conf or
dahdi.conf has a context of from-pstn. The key to actually solving this will
be for you to give us say 10 lines of CLI output.
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[asterisk-users] s extension not working
Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten => s,1,Answer() same => n,Playback(hello-world) same => n,Hangup() Working... [from-pstn] exten => _,1,Answer() same => n,Playback(hello-world) same => n,Hangup() -S -- _ -- 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] [SOLVED] Back-to-back asterisk PRI issue
After following changes my D-Channel comes up and its working!!! :) vi /etc/wanpipe/wanpipe*.conf TDMV_DCHAN = 0 TDMV_HWEC = NO @Thanks all of them who helped here... No beer for others ;) -S From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 23:44:31 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Check out this https://issues.asterisk.org/view.php?id=17270 > From: tilgh...@meg.abyt.es > To: asterisk-users@lists.digium.com > Date: Fri, 25 Mar 2011 17:23:28 -0500 > Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue > > On Friday 25 March 2011 16:23:27 satish patel wrote: > > I just start "Pri set debug on span 1" and its showing D-channel is > > down > > How do you have the underlying T1 signalling set up in > /etc/dahdi/system.conf (on both ends)? > > -- > Tilghman > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Back-to-back asterisk PRI issue
Check out this https://issues.asterisk.org/view.php?id=17270 > From: tilgh...@meg.abyt.es > To: asterisk-users@lists.digium.com > Date: Fri, 25 Mar 2011 17:23:28 -0500 > Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue > > On Friday 25 March 2011 16:23:27 satish patel wrote: > > I just start "Pri set debug on span 1" and its showing D-channel is > > down > > How do you have the underlying T1 signalling set up in > /etc/dahdi/system.conf (on both ends)? > > -- > Tilghman > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Back-to-back asterisk PRI issue
Both server has same content in system.conf file. satish@shirley:~$ cat /etc/dahdi/system.conf # Global data loadzone= us defaultzone = us span = 1,1,0,esf,b8zs bchan = 1-23 dchan=24 echocanceller = mg2,1-23 > From: tilgh...@meg.abyt.es > To: asterisk-users@lists.digium.com > Date: Fri, 25 Mar 2011 17:23:28 -0500 > Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue > > On Friday 25 March 2011 16:23:27 satish patel wrote: > > I just start "Pri set debug on span 1" and its showing D-channel is > > down > > How do you have the underlying T1 signalling set up in > /etc/dahdi/system.conf (on both ends)? > > -- > Tilghman > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Back-to-back asterisk PRI issue
I just start "Pri set debug on span 1" and its showing D-channel is down
satish-desktop*CLI> pri show span
Usage: pri show span
Displays PRI Information on a given PRI span
satish-desktop*CLI> pri show span 1
Primary D-channel: 24
Status: Down, Active
Switchtype: Q.SIG switch
Type: Network
Overlap Dial: 0
Logical Channel Mapping: 0
Timer and counter settings:
N200: 3
N202: 3
K: 7
T200: 1000
T202: 1
T203: 1
T303: 4000
T305: 3
T308: 4000
T309: 6000
T313: 4000
T-HOLD: 4000
T-RETRIEVE: 4000
T-RESPONSE: 4000
Overlap Recv: No
satish-desktop*CLI> pri set debug on span 1
1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state
5(Awaiting establishment)
1 Changing from state 5(Awaiting establishment) to 4(TEI assigned)
1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3)
1 TEI=0 Sending SABME
1 Changing from state 4(TEI assigned) to 5(Awaiting establishment)
Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state
5(Awaiting establishment)
1 Changing from state 5(Awaiting establishment) to 4(TEI assigned)
1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3)
1 TEI=0 Sending SABME
1 Changing from state 4(TEI assigned) to 5(Awaiting establishment)
Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state
5(Awaiting establishment)
1 Changing from state 5(Awaiting establishment) to 4(TEI assigned)
1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3)
1 TEI=0 Sending SABME
1 Changing from state 4(TEI assigned) to 5(Awaiting establishment)
Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state
5(Awaiting establishment)
1 Changing from state 5(Awaiting establishment) to 4(TEI assigned)
1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3)
1 TEI=0 Sending SABME
1 Changing from state 4(TEI assigned) to 5(Awaiting establishment)
Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state
5(Awaiting establishment)
1 Changing from state 5(Awaiting establishment) to 4(TEI assigned)
1 TEI=0 DL event: Q931_DL_EVENT_DL_RELEASE_IND(3)
1 TEI=0 Sending SABME
1 Changing from state 4(TEI assigned) to 5(Awaiting establishment)
Span: 1 Processing event: PRI_EVENT_DCHAN_DOWN
1 TEI=0 Sending SABME
1 TEI=0 Sending SABME
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 21:13:34 +
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
sometime i am getting following error also. what is this means?
[Mar 25 17:11:52] WARNING[5961]: sig_pri.c:985 pri_find_dchan: Span 1: No
D-channels available! Using Primary channel as D-channel anyway!
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 21:04:45 +
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Only Difference is one side card is ECHO Cancellation supported and other is
non-ECHO cancellation. Is there any issue ?
@Asterisk1
Sangoma A102 (non-ECHW)
@Asterisk2
Sangoma A102D (ECHW)
-Satish
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 20:41:09 +
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Okay! i have changed context at master side
; Span 1: WPT1/0 "wanpipe1 card 0" (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-internal
group = 1,24
echocancel = yes
signalling = pri_net
channel => 1-23
Same error nothing change..
satish-desktop*CLI> core set verbose 10
Verbosity was 0 and is now 10
satish-desktop*CLI> core set debug 999
Core debug was 0 and is now 999
== Using SIP RTP CoS mark 5
-- Executing [7527@from-sip:1] Dial("SIP/7623-", "DAHDI/g1/527") in
new stack
[Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION'
> From: tilgh...@meg.abyt.es
> To: asterisk-users@lists.digium.com
> Date: Fri, 25 Mar 2011 15:35:21 -0500
> Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
>
> On Friday 25 March 2011 14:40:40 satish patel wrote:
> > Following is my scenario to connect back to back PRI of two asterisk
> > server. PRI cards are Sangoma A102D
> >
> > [Asterisk1][PRI]-Cross Cable-[Asterisk2]
> >
> > Asteri
Re: [asterisk-users] Back-to-back asterisk PRI issue
sometime i am getting following error also. what is this means?
[Mar 25 17:11:52] WARNING[5961]: sig_pri.c:985 pri_find_dchan: Span 1: No
D-channels available! Using Primary channel as D-channel anyway!
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 21:04:45 +
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Only Difference is one side card is ECHO Cancellation supported and other is
non-ECHO cancellation. Is there any issue ?
@Asterisk1
Sangoma A102 (non-ECHW)
@Asterisk2
Sangoma A102D (ECHW)
-Satish
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 20:41:09 +
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Okay! i have changed context at master side
; Span 1: WPT1/0 "wanpipe1 card 0" (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-internal
group = 1,24
echocancel = yes
signalling = pri_net
channel => 1-23
Same error nothing change..
satish-desktop*CLI> core set verbose 10
Verbosity was 0 and is now 10
satish-desktop*CLI> core set debug 999
Core debug was 0 and is now 999
== Using SIP RTP CoS mark 5
-- Executing [7527@from-sip:1] Dial("SIP/7623-", "DAHDI/g1/527") in
new stack
[Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION'
> From: tilgh...@meg.abyt.es
> To: asterisk-users@lists.digium.com
> Date: Fri, 25 Mar 2011 15:35:21 -0500
> Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
>
> On Friday 25 March 2011 14:40:40 satish patel wrote:
> > Following is my scenario to connect back to back PRI of two asterisk
> > server. PRI cards are Sangoma A102D
> >
> > [Asterisk1][PRI]-Cross Cable-[Asterisk2]
> >
> > Asterisk1
> >
> > ; Span 1 (MASTER)
> > switchtype = national ; commonly referred to as NI2
> > context = from-pstn
> > group = 0,24
> > echocancel = yes
> > signalling = pri_net
> > channel => 1-23
> >
> >
> > Asterisk2
> >
> > ; Span 1
> > switchtype = national ; commonly referred to as NI2
> > context = from-pstn
> > group = 0,24
> > echocancel = yes
> > signalling = pri_cpe
> > channel => 1-23
>
> Here's one confusing part. You're saying that calls that come from the
> master to the slave end up in context from-pstn (on the slave), but calls
> from the slave to the master ALSO end up in from-pstn (on the master).
> Seems like one of them should be "from-internal" or the like. I'm sure
> some of your problem emanate from these settings.
>
> > satish-desktop*CLI>
> > [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable
> > to create channel of type 'DAHDI' (cause 34 - Circuit/channel
> > congestion)
>
> Check the other side for error messages.
>
> > [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor:
> > Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115
> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned
> > -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115
> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned
> > -1: Invalid argument
>
> This problem is due to a misconfiguration. Asterisk cannot handle the local
> network being addressed as the 0.0.0.0 network. You need to use the full
> local address.
>
> --
> Tilghman
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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Re: [asterisk-users] Back-to-back asterisk PRI issue
Only Difference is one side card is ECHO Cancellation supported and other is
non-ECHO cancellation. Is there any issue ?
@Asterisk1
Sangoma A102 (non-ECHW)
@Asterisk2
Sangoma A102D (ECHW)
-Satish
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 20:41:09 +
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Okay! i have changed context at master side
; Span 1: WPT1/0 "wanpipe1 card 0" (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-internal
group = 1,24
echocancel = yes
signalling = pri_net
channel => 1-23
Same error nothing change..
satish-desktop*CLI> core set verbose 10
Verbosity was 0 and is now 10
satish-desktop*CLI> core set debug 999
Core debug was 0 and is now 999
== Using SIP RTP CoS mark 5
-- Executing [7527@from-sip:1] Dial("SIP/7623-", "DAHDI/g1/527") in
new stack
[Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION'
> From: tilgh...@meg.abyt.es
> To: asterisk-users@lists.digium.com
> Date: Fri, 25 Mar 2011 15:35:21 -0500
> Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
>
> On Friday 25 March 2011 14:40:40 satish patel wrote:
> > Following is my scenario to connect back to back PRI of two asterisk
> > server. PRI cards are Sangoma A102D
> >
> > [Asterisk1][PRI]-Cross Cable-[Asterisk2]
> >
> > Asterisk1
> >
> > ; Span 1 (MASTER)
> > switchtype = national ; commonly referred to as NI2
> > context = from-pstn
> > group = 0,24
> > echocancel = yes
> > signalling = pri_net
> > channel => 1-23
> >
> >
> > Asterisk2
> >
> > ; Span 1
> > switchtype = national ; commonly referred to as NI2
> > context = from-pstn
> > group = 0,24
> > echocancel = yes
> > signalling = pri_cpe
> > channel => 1-23
>
> Here's one confusing part. You're saying that calls that come from the
> master to the slave end up in context from-pstn (on the slave), but calls
> from the slave to the master ALSO end up in from-pstn (on the master).
> Seems like one of them should be "from-internal" or the like. I'm sure
> some of your problem emanate from these settings.
>
> > satish-desktop*CLI>
> > [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable
> > to create channel of type 'DAHDI' (cause 34 - Circuit/channel
> > congestion)
>
> Check the other side for error messages.
>
> > [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor:
> > Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115
> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned
> > -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115
> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned
> > -1: Invalid argument
>
> This problem is due to a misconfiguration. Asterisk cannot handle the local
> network being addressed as the 0.0.0.0 network. You need to use the full
> local address.
>
> --
> Tilghman
>
> --
> _
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Back-to-back asterisk PRI issue
Okay! i have changed context at master side
; Span 1: WPT1/0 "wanpipe1 card 0" (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-internal
group = 1,24
echocancel = yes
signalling = pri_net
channel => 1-23
Same error nothing change..
satish-desktop*CLI> core set verbose 10
Verbosity was 0 and is now 10
satish-desktop*CLI> core set debug 999
Core debug was 0 and is now 999
== Using SIP RTP CoS mark 5
-- Executing [7527@from-sip:1] Dial("SIP/7623-", "DAHDI/g1/527") in
new stack
[Mar 25 16:39:47] WARNING[5336]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/7623-' status is 'CONGESTION'
> From: tilgh...@meg.abyt.es
> To: asterisk-users@lists.digium.com
> Date: Fri, 25 Mar 2011 15:35:21 -0500
> Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
>
> On Friday 25 March 2011 14:40:40 satish patel wrote:
> > Following is my scenario to connect back to back PRI of two asterisk
> > server. PRI cards are Sangoma A102D
> >
> > [Asterisk1][PRI]-Cross Cable-[Asterisk2]
> >
> > Asterisk1
> >
> > ; Span 1 (MASTER)
> > switchtype = national ; commonly referred to as NI2
> > context = from-pstn
> > group = 0,24
> > echocancel = yes
> > signalling = pri_net
> > channel => 1-23
> >
> >
> > Asterisk2
> >
> > ; Span 1
> > switchtype = national ; commonly referred to as NI2
> > context = from-pstn
> > group = 0,24
> > echocancel = yes
> > signalling = pri_cpe
> > channel => 1-23
>
> Here's one confusing part. You're saying that calls that come from the
> master to the slave end up in context from-pstn (on the slave), but calls
> from the slave to the master ALSO end up in from-pstn (on the master).
> Seems like one of them should be "from-internal" or the like. I'm sure
> some of your problem emanate from these settings.
>
> > satish-desktop*CLI>
> > [Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable
> > to create channel of type 'DAHDI' (cause 34 - Circuit/channel
> > congestion)
>
> Check the other side for error messages.
>
> > [Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor:
> > Cannot connect [Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115
> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned
> > -1: Invalid argument [Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115
> > __sip_xmit: sip_xmit of 0x14249d0 (len 763) to 0.0.29.103:5060 returned
> > -1: Invalid argument
>
> This problem is due to a misconfiguration. Asterisk cannot handle the local
> network being addressed as the 0.0.0.0 network. You need to use the full
> local address.
>
> --
> Tilghman
>
> --
> _
> -- 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
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Re: [asterisk-users] Back-to-back asterisk PRI issue
One more thing i would like to tell you i have following wanpipe configuration at both side @Asterisk1 root@satish-desktop:~# cat /etc/wanpipe/wanpipe1.conf | grep -i clock TE_CLOCK= MASTER TE_REF_CLOCK= 0 @Asterisk2 root@shirley:/# cat /etc/wanpipe/wanpipe2.conf | grep -i clock TE_CLOCK= NORMAL TE_REF_CLOCK= 0 From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 20:25:31 + Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue Thanks Doug, I tried that also but result is same. > Date: Fri, 25 Mar 2011 16:11:49 -0400 > From: supp...@drdos.info > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue > > satish patel wrote: > > group = 0,24 > > Granted, I'm still running 1.4.x, but I don't believe the above is valid. > > My guess is it should be: > > group = 0 > > Doug > > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Back-to-back asterisk PRI issue
Thanks Doug, I tried that also but result is same. > Date: Fri, 25 Mar 2011 16:11:49 -0400 > From: supp...@drdos.info > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue > > satish patel wrote: > > group = 0,24 > > Granted, I'm still running 1.4.x, but I don't believe the above is valid. > > My guess is it should be: > > group = 0 > > Doug > > > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Back-to-back asterisk PRI issue
Asterisk1
satish-desktop*CLI> dahdi show status
Description Alarms IRQbpviol CRC4 Fra Codi
Options LBO
wanpipe1 card 0 OK 0 0 0 ESF B8ZS
0 db (CSU)/0-133 feet (DSX-1)
wanpipe2 card 1 UNCONFI 0 0 0 CAS Unk
0 db (CSU)/0-133 feet (DSX-1)
satish-desktop*CLI>
Asterisk2
shirley*CLI> dahdi show status
Description Alarms IRQbpviol CRC4 Fra Codi
Options LBO
wanpipe1 card 0 OK 0 0 0 ESF B8ZS
0 db (CSU)/0-133 feet (DSX-1)
wanpipe2 card 1 RED 0 0 0 CAS Unk
0 db (CSU)/0-133 feet (DSX-1)
shirley*CLI>
From: will...@stillwellsoft.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 16:04:12 -0400
Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
Did you check so see if the pri is up? Also, make sure wanpipe & dahdi is setup
correctly. From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, March 25, 2011 3:41 PM
To: asterisk-users
Subject: [asterisk-users] Back-to-back asterisk PRI issue Following is my
scenario to connect back to back PRI of two asterisk server. PRI cards are
Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_net
channel => 1-23
Asterisk2
; Span 1
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_cpe
channel => 1-23
Following is my extensions.conf stuff on both machine (extension number could
be change)
[from-pstn]
exten => s,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
[from-sip]
exten => _7XXX,1,Answer()
same => n,Dial(SIP/${EXTEN})
same => n,Hangup()
exten => 7527,1,Dial(DAHDI/G0/7527)
But i am getting following error when i am calling from A to B
satish-desktop*CLI>
[Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
[Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect
[Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
[Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
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Re: [asterisk-users] [SOLVED] reload command not availeble asterisk 1.8.x
No kidding.. found this line second server. Thanks!! root@shirley:/# cat /etc/asterisk/modules.conf | grep res_clialiases.so noload => res_clialiases.so From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 25 Mar 2011 19:53:58 + Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x satish-desktop*CLI> module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 res_clialiases.so CLI Aliases 0 2 modules loaded shirley*CLI> module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 1 modules loaded > Date: Fri, 25 Mar 2011 15:45:13 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x > > On 11-03-25 03:13 PM, satish patel wrote: > > > > Both servers files are identical.. > > > *CLI> module show like cli > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] reload command not availeble asterisk 1.8.x
satish-desktop*CLI> module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 res_clialiases.so CLI Aliases 0 2 modules loaded shirley*CLI> module show like cli Module Description Use Count res_clioriginate.soCall origination and redirection from th 0 1 modules loaded > Date: Fri, 25 Mar 2011 15:45:13 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x > > On 11-03-25 03:13 PM, satish patel wrote: > > > > Both servers files are identical.. > > > *CLI> module show like cli > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Back-to-back asterisk PRI issue
Following is my scenario to connect back to back PRI of two asterisk server.
PRI cards are Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_net
channel => 1-23
Asterisk2
; Span 1
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel = yes
signalling = pri_cpe
channel => 1-23
Following is my extensions.conf stuff on both machine (extension number could
be change)
[from-pstn]
exten => s,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
[from-sip]
exten => _7XXX,1,Answer()
same => n,Dial(SIP/${EXTEN})
same => n,Hangup()
exten => 7527,1,Dial(DAHDI/G0/7527)
But i am getting following error when i am calling from A to B
satish-desktop*CLI>
[Mar 25 15:40:19] WARNING[4519]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
[Mar 25 15:40:19] WARNING[4519]: acl.c:698 ast_ouraddrfor: Cannot connect
[Mar 25 15:40:19] WARNING[4519]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
[Mar 25 15:40:19] WARNING[4371]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x14249d0 (len 763) to 0.0.29.103:5060 returned -1: Invalid argument
--
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Re: [asterisk-users] reload command not availeble asterisk 1.8.x
Both servers files are identical.. root@satish-desktop:~# cat /etc/asterisk/cli_aliases.conf | grep reload reload=module reload ; Alias for making voicemail reload actually do module reload app_voicemail.so ;voicemail reload=module reload app_voicemail.so ; This will make the CLI command "mr" behave as though it is "module reload". ;mr=module reload extensions reload=dialplan reload root@shirley:/# cat /etc/asterisk/cli_aliases.conf | grep reload reload=module reload ; Alias for making voicemail reload actually do module reload app_voicemail.so ;voicemail reload=module reload app_voicemail.so ; This will make the CLI command "mr" behave as though it is "module reload". ;mr=module reload extensions reload=dialplan reload > Date: Fri, 25 Mar 2011 14:57:14 -0400 > From: pabelan...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] reload command not availeble asterisk 1.8.x > > On 11-03-25 02:49 PM, satish patel wrote: > > I have two asterisk 1.8.3.2 same version on both machine but why one > > asterisk has "reload" command but other doesn't ? > > > *CLI> module reload > > 'reload' is no longer a valid command. I'm guess one box has > cli_aliases.conf, while the other does not. > > -- > Paul Belanger > Digium, Inc. | Software Developer > twitter: pabelanger | IRC: pabelanger (Freenode) > Check us out at: http://digium.com & http://asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reload command not availeble asterisk 1.8.x
Hey Guys! I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has "reload" command but other doesn't ? satish-desktop*CLI> core show version Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on 2011-03-25 16:10:39 UTC satish-desktop*CLI> re realtime reload shirley*CLI> core show version Asterisk 1.8.3.2 built by root @ shirley on a x86_64 running Linux on 2011-03-22 18:38:19 UTC shirley*CLI> re destroy load mysqlstoreupdate update2 -- _ -- 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] dahdi restart warning
AHH! wait a min.. look like i figured out these thing i found inside following file. what those entries for ? root@shirley:/etc/asterisk# cat /etc/asterisk/users.conf | grep -v ';' [general] fullname = New User userbase = 6000 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = yes hasmanager = no callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 24 Mar 2011 18:50:49 + Subject: Re: [asterisk-users] dahdi restart warning dump!! Can anybody please reply me on below email? I did lots of gogling but no clear answer anywhere related below errors. I will appreciate your help. -S From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 23 Mar 2011 21:03:43 + Subject: [asterisk-users] dahdi restart warning What is this error ? is this harmful ? *CLI>*CLI> dahdi restart [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Mar 23 14:01:10] WARNING[4315]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] dahdi restart warning
dump!! Can anybody please reply me on below email? I did lots of gogling but no clear answer anywhere related below errors. I will appreciate your help. -S From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 23 Mar 2011 21:03:43 + Subject: [asterisk-users] dahdi restart warning What is this error ? is this harmful ? *CLI>*CLI> dahdi restart [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Mar 23 14:01:10] WARNING[4315]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi restart warning
What is this error ? is this harmful ? *CLI>*CLI> dahdi restart [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Mar 23 14:01:10] WARNING[4315]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! -- _ -- 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] Sangoma A102D wanpiple issue with dahdi
added: what is this error ? root@shirley:~# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wanpipe: error No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: . From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 23 Mar 2011 17:56:42 + Subject: [asterisk-users] Sangoma A102D wanpiple issue with dahdi Hey Guy, I have ubuntu 10.04 64bit and compiled dahdi / wanpipe 3.5.19 /asterisk-1.8.x I didn't understand what is the relation between wanpipe and dahdi ? do i need to start wanrouter service ? I am getting weird errors and my system got kernel panic many time when i restart dahdi service. any idea ? what is the startup sequence of all these service ? root@example:/etc/asterisk# /etc/init.d/dahdi stop Unloading DAHDI hardware modules: ERROR: Module wanpipe is in use ERROR: Module dahdi_echocan_mg2 is in use ERROR: Module dahdi is in use by dahdi_echocan_mg2,wanpipe error root@example:/etc/asterisk# wanrouter stop Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe2 Shutting down device: wanpipe1 wanconfig: WAN device wanpipe1 did not shutdown : ioctl(wanpipe1,ROUTER_DOWN) failed: : 16 - Device or resource busy If you router was not running ignore this message !! Otherwise, check the /var/log/wanrouter and /var/log/messages for errors Devices Still Running: wanpipe1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A102D wanpiple issue with dahdi
Hey Guy, I have ubuntu 10.04 64bit and compiled dahdi / wanpipe 3.5.19 /asterisk-1.8.x I didn't understand what is the relation between wanpipe and dahdi ? do i need to start wanrouter service ? I am getting weird errors and my system got kernel panic many time when i restart dahdi service. any idea ? what is the startup sequence of all these service ? root@example:/etc/asterisk# /etc/init.d/dahdi stop Unloading DAHDI hardware modules: ERROR: Module wanpipe is in use ERROR: Module dahdi_echocan_mg2 is in use ERROR: Module dahdi is in use by dahdi_echocan_mg2,wanpipe error root@example:/etc/asterisk# wanrouter stop Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe2 Shutting down device: wanpipe1 wanconfig: WAN device wanpipe1 did not shutdown : ioctl(wanpipe1,ROUTER_DOWN) failed: : 16 - Device or resource busy If you router was not running ignore this message !! Otherwise, check the /var/log/wanrouter and /var/log/messages for errors Devices Still Running: wanpipe1 -- _ -- 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] Sangoma wapipe installation error
Hey, I did ./Setup dahdi and everything went well but i didn't find any
command wancfg_dahdi
--
WANPIPE v3.5.19 Installation Script
Copyright (c) 1995-2010, Sangoma Technologies Inc.
--
WANPIPE META CONFIGURATION
There are two configuration files associated with WANPIPE.
1) /usr/local/src/asterisk/wanpipe-3.5.19/wanrouter.rc:
- defines locations of important files such as lock
and configuration files as well as start/stop
order of multiple WANPIPE devices.
2) /usr/local/src/asterisk/wanpipe-3.5.19/wanpipe1.conf:
- main configuration file for each WANPIPE device.
- defines interfaces, hardware and protocol information.
- this file can be created using the 'wancfg' GUI
utility or manually based on sample files located
in /etc/wanpipe/samples.
Please read the WanpipeInstallation.(pdf/txt) manual for further
information.
Wanpipe META config file found in /etc/wanpipe directory
Wanpipe startup sequence: wanpipe1
--
WANPIPE v3.5.19 Installation Script
Copyright (c) 1995-2010, Sangoma Technologies Inc.
--
WANPIPE UTILITIES SETUP
WANPIPE utilities are used to:
1) create configuration files: for Zaptel and Asterisk
/usr/sbin/wancfg_zaptel #Zaptel and Asterisk
/usr/sbin/wancfg_dahdi #Dahdi and Asterisk
/usr/sbin/wancfg_smg#BRI/SS7, Zaptel and Asterisk
/usr/sbin/wancfg_tdmapi #TDM API
2) create WANPIPE WAN/IP configuration files.
(/usr/sbin/wancfg)
3) start,stop,restart individual/all devices and interfaces.
(/usr/sbin/wanrouter)
4) debug line, protocol and driver problems.
(/usr/sbin/wanpipemon)
5) aid in WANPIPE API development
(/etc/wanpipe/api)
Refer to the WanpipeInstallation.(pdf/txt) for more information.
Compiling WANPIPE Utilities ... Done.
Compiling WANPIPE WanCfg Utility ...Done.
Compiling WANPIPE LibSangoma API library ...Done.
Compiling WANPIPE LibStelephony API library ...Done.
Compiling WANPIPE API Development Utilities ...Done.
Compiling WANPIPE HWEC Utilities ...Done.
WANPIPE Environment Setup Complete !!!
Installing WANPIPE Files ... !
Installing WANPIPE Utilities in /usr/sbin
Installing wanrouter.rc in /etc/wanpipe
Installing wanpipe libraries in /etc/wanpipe
Installing firmware in /etc/wanpipe/firmware
Installing documentation in /usr/share/doc/wanpipe
Installing sample api code in /etc/wanpipe/api
Installing AFT Firmware update utility in /etc/wanpipe/util
cp: cannot overwrite non-directory `/etc/wanpipe/util/wan_aftup' with directory
`util/wan_aftup/'
Installing driver headers in /etc/wanpipe/api/include/linux
Installing Hardware Echo Cancel Utilites
--
WANPIPE v3.5.19 Installation Script
Copyright (c) 1995-2010, Sangoma Technologies Inc.
--
WANPIPE INSTALLATON: COMPLETE
WANPIPE installation is now complete. WANPIPE kernel drivers
and configuration/debug utilities have been compiled and installed.
1) Proceed to configure the WANPIPE drivers:
Asterisk/Zaptel : /usr/sbin/wancfg_zaptel
Asterisk/Dahdi : /usr/sbin/wancfg_dahdi
TDM API : /usr/sbin/wancfg_tdmapi
SMG SS7/BRI/PRI : /usr/sbin/wancfg_smg
WAN Routing/API : /usr/sbin/wancfg
2) Use the /usr/sbin/wanrouter startup script to start and stop
the router. (eg: wanrouter start)
3) To uninstall WANPIPE package run ./Setup remove
Please read http://wiki.sangoma.com for further instructions.
root@example:/usr/local/src/asterisk/wanpipe-3.5.19# wancfg_dahdi
wancfg_dahdi: command not found
root@example:/usr/local/src/asterisk/wanpipe-3.5.19#
Date: Wed, 23 Mar 2011 09:28:25 +0100
From: t...@ovm-group.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sangoma wapipe installation error
Try:
cd /usr/src/dahdi
./Setup dahdi
That's it.
Am 22.03.2011 21:06, schrieb satish patel:
Hey!
I am installing Sangoma A102D wanpipe driver and i got following
error. what is this ? why dir isn't there ?
wanpipe-3.5.16 # make dahdi DAHDI_DIR=/usr/src/dahdi
wanpipe-3.5.16 # make install
Send
Installing Wanpipe Firmware update utility in
/etc/wanpipe/util/wan_aftup
Re: [asterisk-users] Queue pause vs logged out ?
Thanks to everyone who replied on this thread. -- Sent from my iPhone On Mar 22, 2011, at 1:31 PM, Carlos Chavez wrote: On Tue, 2011-03-22 at 15:16 +0100, Lenz Emilitri wrote: Maybe not much from the point of view of queues, but this may make quite a difference from the point of view of monitoring your call-center. :) l. 2011/3/21 satish patel Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish Not just statistics. The one thing I can think of that will affect wether the agent is paused or logged off is the joinempty and leavewhenempty options in queues.conf. The behavior is different if you use yes or strict. Read que example queues.conf to know how it affects you. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma wapipe installation error
Hey! I am installing Sangoma A102D wanpipe driver and i got following error. what is this ? why dir isn't there ? wanpipe-3.5.16 # make dahdi DAHDI_DIR=/usr/src/dahdi wanpipe-3.5.16 # make install Send Installing Wanpipe Firmware update utility in /etc/wanpipe/util/wan_aftup install -D wan_aftup /usr/sbin/wan_aftup install -d /etc/wanpipe/util/wan_aftup/scripts install: cannot create directory `/etc/wanpipe/util/wan_aftup': Not a directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/asterisk/wanpipe-3.5.16/util/wan_aftup' make[1]: *** [install] Error 2 make[1]: Leaving directory `/usr/local/src/asterisk/wanpipe-3.5.16/util' make: *** [install_util] Error 2 -- _ -- 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] Asterisk PRI back-to-back connect
Thanks for reply, We have full PBX running in production and SIP + PRI line. But now i am planing to upgrade my existing server with new hardware server/telephony card etc.. I have build new server with asterisk 1.8.x with new PRI cards now before switch PRI line from old one to new asterisk i want to make sure my new PRI cards are working properly and configured ( I have only 10 min downtime to switch over so i can do testing on that downtime window) Thats why i am planing to connect two asterisk both back to back over PRI line for just **TEST**ing to make sure my PRI cards working and able to handle calls.. -Satish > Date: Tue, 22 Mar 2011 14:05:47 -0400 > From: rswago...@gmail.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Asterisk PRI back-to-back connect > > On Tue, Mar 22, 2011 at 12:53 PM, satish patel wrote: > > Hey Guys! > > > > We have two Asterisk with A102D Sangoma cards now i want to connect them > > back-to-back over PRI line via Cross-cable so what would be the > > configuration specially timing source and all? anybody did it before like > > this ? > > > > I want to make sure everything before putting in production.. (saving my > > downtime) > > > > -S > > > > If is no different then setting up the card to connect with a telco. > One Asterisk box will be the net and the other is cpe. You can use > whatever protocol national, 5ess, etc you like. Any reason not to join > the boxes via SIP? > > Ryan > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PRI back-to-back connect
Hey Guys! We have two Asterisk with A102D Sangoma cards now i want to connect them back-to-back over PRI line via Cross-cable so what would be the configuration specially timing source and all? anybody did it before like this ? I want to make sure everything before putting in production.. (saving my downtime) -S -- _ -- 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] Queue pause vs logged out ?
Thanks Louis, We have only single queue. so is it ok if we use only pause option not logged out ? -Satish > From: carreir...@gmail.com > To: asterisk-users@lists.digium.com > Date: Mon, 21 Mar 2011 12:36:09 -0400 > Subject: Re: [asterisk-users] Queue pause vs logged out ? > > Satish, > > Paused is like "Not Ready" on other systems. The user is logged in but is > working on something else or took a break (e.g. restroom). Calls rotate pass > the user while paused. > > v/r, > Louis > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel > Sent: Monday, March 21, 2011 12:25 PM > To: asterisk-users > Subject: [asterisk-users] Queue pause vs logged out ? > > Hey Guys, > > I knew this is stupid question but i just want to know what is the difference > between Queue member logged out vs Pause ? > > -Satish > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue pause vs logged out ?
Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -- _ -- 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] Executing shell commands via AMI
But what about if asterisk running with non-privilege user? Still it is not a good idea. -- Sent from my iPhone On Mar 16, 2011, at 2:33 PM, Tilghman Lesher wrote: On Wednesday 16 March 2011 13:14:40 Vinícius Fontes wrote: action: command command: ! /bin/ls -l / For security reasons, you cannot do this. This is intentional, not a bug. Consider the command 'rm -rf /' for the reason why. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Solved: Auto Answer in manager
Variable: SIPADDHEADER=Alert-Info: Ring Answer From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 15 Mar 2011 20:59:23 + Subject: [asterisk-users] Auto Answer in manager Hi All, I am doing auto answering call from manager but it seems not working any idea ? following commands i am passing to my manager. My phone only ringing not answering we have asterisk 1.8 Action: Originate Channel: SIP/7527 Context: all-page Priority: 1 Variable: SIPAddHeader Value: Alert-Info: Ring Answer CallerID: System Page Action: Logoff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Answer in manager
Hi All, I am doing auto answering call from manager but it seems not working any idea ? following commands i am passing to my manager. My phone only ringing not answering we have asterisk 1.8 Action: Originate Channel: SIP/7527 Context: all-page Priority: 1 Variable: SIPAddHeader Value: Alert-Info: Ring Answer CallerID: System Page Action: Logoff -- _ -- 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] call file for page auto-call
Thanks for you input but how to do SIPAddHeader(Alert-Info: Ring Answer) for auto answer my polycom phones and how to create group in .call file I am reading at http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out but didn't found anything related group calling. may be i am missing something could point me out.. -S From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 15 Mar 2011 13:11:16 -0500 Subject: Re: [asterisk-users] call file for page auto-call From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Tuesday, March 15, 2011 1:06 PM To: asterisk-users Subject: [asterisk-users] call file for page auto-call Hey Support, I am planing to implement new page system with asterisk 1.8 we have 200 SIP calls and page() will overkill my system if announce in one shot. so i am planing to record and play page over 50...50...50 chunk.. I am planing to do with .call file for auto calling after record message but i don't know how to call multiple extension ? and how to use page() with .call file for auto-answer and auto-call? Appreciate your help.. -S One suggestion – set up 4 “call groups”. Group 1 calls first 50 phones, Group 2 51-100, etc. If you set it up like 601, 602, etc. then in your call file you can test with 101 to get what you want, then change it to 601. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call file for page auto-call
Hey Support, I am planing to implement new page system with asterisk 1.8 we have 200 SIP calls and page() will overkill my system if announce in one shot. so i am planing to record and play page over 50...50...50 chunk.. I am planing to do with .call file for auto calling after record message but i don't know how to call multiple extension ? and how to use page() with .call file for auto-answer and auto-call? Appreciate your help.. -S -- _ -- 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] Asterisk 1.8 paging with ploycom
Hey, Could you give me some idea how to do this ? I meant record and play ? do you want me to use .call file ? -Satish > Date: Mon, 14 Mar 2011 16:29:19 + > From: a...@datavox.co.uk > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > If I was worried I'd record the 'page' first - and then play it back to > 50 handsets at a time (using a loop). > > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish > patel > Sent: 14 March 2011 16:25 > To: asterisk-users > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > > Thanks Kevin, > > I test page application and it works but i am worried about i have 200 > SIP phone. Do you think asterisk page application can handle that number > of page ? > > Just worried about my asterisk. I don't want to crach :( > > -Satish > > > > > Date: Mon, 14 Mar 2011 11:18:36 -0500 > > From: kpflem...@digium.com > > To: asterisk-users@lists.digium.com > > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > > > On 03/14/2011 10:01 AM, satish patel wrote: > > > Hey Guys, > > > > > > I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi > > > stopped working look like asterisk 1.8 did some changes in manager > apps > > > i am doing following.. my phone is ringing but not auto answer could > you > > > give me some issue what i am doing wrong ? > > > > The manager interface has indeed changed between 1.2 and 1.8 (likely > it > > has changed many times), and you would do yourself a world of good to > > read through the upgrade notes that came with Asterisk 1.8 to > understand > > how you might need to change your scripts. > > > > In addition, Asterisk 1.8 has a built-in Page() application you can > use > > from the dialplan to achieve what it appears you were trying to > achieve > > with your AGI script. > > > > -- > > Kevin P. Fleming > > Digium, Inc. | Director of Software Technologies > > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: > kpfleming > > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > > Check us out at www.digium.com & www.asterisk.org > > > > -- > > _ > > -- 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 > > > If you have received this communication in error we would appreciate > you advising us either by telephone or return of e-mail. The contents > of this message, and any attachments, are the property of DataVox, > and are intended for the confidential use of the named recipient only. > If you are not the intended recipient, employee or agent responsible > for delivery of this message to the intended recipient, take note that > any dissemination, distribution or copying of this communication and > its attachments is strictly prohibited, and may be subject to civil or > criminal action for which you may be liable. > Every effort has been made to ensure that this e-mail or any attachments > are free from viruses. While the company has taken every reasonable > precaution to minimise this risk, neither company, nor the sender can > accept liability for any damage which you sustain as a result of viruses. > It is recommended that you should carry out your own virus checks > before opening any attachments. > > Registered in England. No. 27459085. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 paging with ploycom
We don't have multicast network configuration in our LAN :( From: steve-li...@geekinter.net Date: Mon, 14 Mar 2011 16:29:55 + To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom On 14 Mar 2011, at 16:24, satish patel wrote:I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Do they support multicast? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 paging with ploycom
Thanks Kevin, I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Just worried about my asterisk. I don't want to crach :( -Satish > Date: Mon, 14 Mar 2011 11:18:36 -0500 > From: kpflem...@digium.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > On 03/14/2011 10:01 AM, satish patel wrote: > > Hey Guys, > > > > I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi > > stopped working look like asterisk 1.8 did some changes in manager apps > > i am doing following.. my phone is ringing but not auto answer could you > > give me some issue what i am doing wrong ? > > The manager interface has indeed changed between 1.2 and 1.8 (likely it > has changed many times), and you would do yourself a world of good to > read through the upgrade notes that came with Asterisk 1.8 to understand > how you might need to change your scripts. > > In addition, Asterisk 1.8 has a built-in Page() application you can use > from the dialplan to achieve what it appears you were trying to achieve > with your AGI script. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 paging with ploycom
Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root@ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk Call Manager/1.1 Action: Login Username: allpage Secret: xxx Events: off Action: Originate Channel: SIP/7527 Context: all-page Priority: 1 Variable: SIPADDHEADER="Call-Info: sip:172.30.254.211" Variable: ALERT_INFO="Ring Answer" Extension: CallerID: System Page Action: Logoff Here my phone SIP/7527 is ringing but not auto answer. why ? -- _ -- 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] Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
Hey Steve,
I got following error when i change AGI to System
ubuntu-test*CLI>
== Using SIP RTP CoS mark 5
-- Executing [7770@from-sip:1] System("SIP/7623-0029",
"/var/lib/asterisk/agi-bin/allpage.agi") in new stack
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
== Using SIP RTP CoS mark 5
[Mar 13 09:43:36] NOTICE[18985]: channel.c:5167 __ast_request_and_dial: Unable
to request channel SIP/7657
[Mar 13 09:43:36] ERROR[18985]: utils.c:1177 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Mar 13 09:43:36] ERROR[18985]: utils.c:1177 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Mar 13 09:43:36] ERROR[18985]: utils.c:1177 ast_careful_fwrite: fwrite()
returned error: Broken pipe
-- Executing [7770@from-sip:2] MeetMe("SIP/7623-0029", "7770,dq") in
new stack
-- Created MeetMe conference 1023 for conference '7770'
-- Hungup 'DAHDI/pseudo-252942591'
== Spawn extension (from-sip, 7770, 2) exited non-zero on 'SIP/7623-0029'
[Mar 13 09:43:55] ERROR[18918]: utils.c:1177 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Mar 13 09:43:55] ERROR[18918]: utils.c:1177 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Mar 13 09:43:55] ERROR[18918]: utils.c:1177 ast_careful_fwrite: fwrite()
returned error: Broken pipe
> Date: Fri, 11 Mar 2011 13:58:43 -0800
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1.8 AGI error ast_carefulwrite:
> write() returned error
>
> On Fri, 11 Mar 2011, satish patel wrote:
>
> > We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script
> > doesn't working We have allpage.agi script for paging system on all
> > polycom 501 but after upgrade it broke. Any idea what is this error ?
>
> > [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write()
> > returned error: Broken pipe
>
> Without source code, I'd guess you are violation the AGI protocol.
>
> What language are you using?
>
> which AGI library are you using?
>
> Can you reduce your source code to a simple application that reliably
> reproduces
> the error.
>
> Can you post the source to the simplified application?
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- 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
--
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Re: [asterisk-users] Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
AH!! Boy I didn't notice that it's not AGI. But In 1.2 it's working with AGI apps. When i am running it on bash it excute successfully and ringing phone but no auto answer working. Let me try with system applications and I will let you know. Thanks for helping me with this. -- Sent from my iPhone On Mar 11, 2011, at 8:45 PM, Steve Edwards wrote: Un-top-posting... On Fri, 11 Mar 2011, satish patel wrote: We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned error: Broken pipe On Mar 11, 2011, at 4:58 PM, Steve Edwards wrote: Without source code, I'd guess you are violation the AGI protocol. Can you reduce your source code to a simple application that reliably reproduces the error. Can you post the source to the simplified application? On Fri, 11 Mar 2011, Satish Patel wrote: I am not in office so i can't post script right now but will so once reach home. If you want to take a look at script I have following URL where someone already doing discusion. My script is pretty similer but I am grabbing all active extension via asterisk CLI commands not statically hardcoded. http://www.freepbx.org/forum/freepbx/tips-and-tricks/delayed-paging If you are referring to the allpage.agi script posted about 40% down the page... It is not an AGI. Note that it does not use any AGI library and that it does not read the AGI environment from STDIN -- which violates the AGI protocol. The allpage script connects to Asterisk via TCP using the AMI protocol. In your dialplan, if you change 'agi(allpage.agi)' to 'system (allpage.agi)' does it behave as you expect? Can you execute the script from a shell command line? -- Thanks in advance, --- -- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
Thanks for reply Steve, I am not in office so i can't post script right now but will so once reach home. By the way that script working great in asterisk 1.2 my production machine. But now I'm testing on 1.8.x and having issue which I mentioned before. This script is perl script and it going to grab all active sip extension and using manager to call all poycom phone via Ring Anwer sipheader. If you want to take a look at script I have following URL where someone already doing discusion. My script is pretty similer but I am grabbing all active extension via asterisk CLI commands not statically hardcoded. http://www.freepbx.org/forum/freepbx/tips-and-tricks/delayed-paging -- Sent from my iPhone On Mar 11, 2011, at 4:58 PM, Steve Edwards wrote: On Fri, 11 Mar 2011, satish patel wrote: We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? [Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write () returned error: Broken pipe Without source code, I'd guess you are violation the AGI protocol. What language are you using? which AGI library are you using? Can you reduce your source code to a simple application that reliably reproduces the error. Can you post the source to the simplified application? -- Thanks in advance, --- -- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
Hey Guys,
We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't
working We have allpage.agi script for paging system on all polycom 501 but
after upgrade it broke. Any idea what is this error ?
extension.conf
exten => 7770,1,agi(allpage.agi)
exten => 7770,2,meetme(7770,dq)
exten => 7770,3,playback(beep)
exten => 7770,4,hangup
following is agi debug
-- Executing [7770@from-sip:1] AGI("SIP/7657-0015", "allpage.agi") in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/allpage.agi
AGI Tx >> agi_request: allpage.agi
AGI Tx >> agi_channel: SIP/7657-0015
AGI Tx >> agi_language: en
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: 1299876046.29
AGI Tx >> agi_version: 1.8.2.3
AGI Tx >> agi_callerid: 7657
AGI Tx >> agi_calleridname: iPhone
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 7770
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: from-sip
AGI Tx >> agi_extension: 7770
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >> agi_threadid: -1345438864
AGI Tx >>
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Remote UNIX connection
-- Remote UNIX connection disconnected
AGI Rx << VERBOSE "Found extension (None) in use." 1
allpage.agi: Found extension (None) in use.
AGI Tx >> 200 result=1
AGI Rx << VERBOSE "Found extension 7657 in use." 1
allpage.agi: Found extension 7657 in use.
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "Adding extension 7527 to call list" 1
allpage.agi: Adding extension 7527 to call list
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "Adding extension 7623 to call list" 1
allpage.agi: Adding extension 7623 to call list
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "NOT Adding extension 7657 to call list" 2
== allpage.agi: NOT Adding extension 7657 to call list
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
AGI Rx << VERBOSE "Doing 7527" 0
allpage.agi: Doing 7527
AGI Tx >> 200 result=1
[Mar 11 15:40:46] ERROR[3140]: utils.c:1130 ast_carefulwrite: write() returned
error: Broken pipe
-- AGI Script allpage.agi completed, returning 0
-- Executing [7770@from-sip:2] MeetMe("SIP/7657-0015", "7770,dq") in
new stack
-- Created MeetMe conference 1023 for conference '7770'
-- Hungup 'DAHDI/pseudo-729745277'
== Spawn extension (from-sip, 7770, 2) exited non-zero on 'SIP/7657-0015'
--
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Re: [asterisk-users] One Way Audio
you I have also tried those settings. The main thing is coming from my voip provider all I am doing is bridging the calls to two other devices (1 trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with an IVR and when I call in I can not hear the IVR. However if I call directly to a SIP client the person answering the SIP phone can hear me but I can not hear them at all. Its definately not a NAT issue which is what makes it even more confusing. When the call is in place a sip show channels shows me both lefs of the call and they are both using either alaw or ulaw so it should not be a codec translation issue either. On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel wrote: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] One Way Audio
What about your sip clients? Are they on public network? Try on sip.conf Nat=no/yes conreinvite=yes/no -- Sent from my iPhone On Mar 9, 2011, at 6:11 PM, Tim King wrote: IPTBALES is off and I have all firewalls disabled. All network elements currently involved have public IP's assigned to them. My main asterisk box has a public IP. I have multiple trunks to voip peers for inbound and outbound calls which are also all public IP's. My two clients are trunked via IAX and one is a Trixbox and the other is a digium AA50 which both also have public IP's assigned to them. On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime wrote: How is your network is organized? Is your server behind a firewal, about iptables ? On Wed, Mar 9, 2011 at 5:40 PM, Tim King wrote: I am having trouble with no return audio on inbound calls. I have been working on this for 18 hours and even built a fresh server and moved everything over and I am getting the same results. I need someone that can help get this resolved tonight. I can provide access to all machines involved. Please email me at tim.compnetw...@gmail.com if you can help. -- _ -- 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 -- Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk pri card replecement
Hey guys, Currently we have non HWEC sangoma pri card but now we are planing to replace card with HWEC support card for echo cancellation. So in this case do I need to re-install everything? Like zaptel, asterisk etc.. Or just replace the card? -- Sent from my iPhone -- _ -- 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] [ASK] can't make call
There is no issue between OS and asterisk. Asterisk is compatible with any linux distribution - So there is no problem. Post some logs / config of sip.conf / extension.conf etc.. make sure your sip clients are registers on asterisk run following command on asterisk CLI >sip show peers or >sip show peer -satish Date: Thu, 3 Mar 2011 08:07:03 -0800 From: don_ba...@yahoo.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] [ASK] can't make call hi, i'm newbie in asterisk n this mailing list i just trying to make this asterisk server for call,n i can't make it.. my asterisk server just respons with REGISTER n SUBSCRIBE method i,m using asterisk 1.4.17 in ubuntu 8.04 with lan connection and just 2 client there.is there a problem with compatible on the series of the ubuntu and the asterisk??? *i'm sorry about my english..:D best regard sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma PCI vs PCI Express card
Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. -Satish -- _ -- 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] VoIP Bandwidth Calculator
I'm using iftop command in Linux and it pretty good though. -- Sent from my iPhone On Mar 3, 2011, at 6:34 AM, "Faisal Hanif" wrote: You can find lots by googling but none can give realtime stats as it depends on network. Packet drop, retransmit, codec type will make lot of vibrations From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, March 03, 2011 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoIP Bandwidth Calculator Hi, Does anyone have a good VoIP Bandwidth Calculator? Thanks Dan Journo Kesher Communications (UK) Business Phone Systems | Hosted PBX -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wav files are not playing asterisk
Do you have complied wav file support in asterisk? -- Sent from my iPhone On Mar 1, 2011, at 9:11 AM, "Danny Nicholas" wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Tuesday, March 01, 2011 3:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] wav files are not playing asterisk Hi I try to play a wav file in asterisk ,but its accepting only gsm files.Do u know where I need to change to make it works. Thanks Nikhil -- If you have foo.gsm and foo.wav, foo.gsm will always play. Rename foo.gsm to foo1.gsm and foo.wav should play. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
