[asterisk-users] Segmentation Fault
Hey, I have installed Asterisk 1.8 on slackware 13.1, php, mysql and apache. I am later to install freepbx to help with reporting on VOIP activity. However, after installing asterisk, I am getting a segment fault. My log file shows this: darkstar kernel: asterisk[2660]: segfault at 81c4f ip 77514810 sp 7fffcd48 error 4 in libc-2.11.1.so[77492000+16b000] I have used gdb so that I can perform a backtrace however the program executes and exits without a stack thus not helpful. Any help is appreciated! Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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 reload, to execute file
Say, When * reloads it changes the file permissions of below file. How can I call an executable which corrects for this? chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi Tx Shaun -- _ -- 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] dundi
Dear is it possible to send ring(call) to all devices with same (sip_username) in all servers ? in this schematics, some bodies have shared lines. so all lines must be in service . Best -- Pezhman Lali -- _ -- 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] *8 pickup not releasing channel
Hi, At a site running Asterisk 1.6.2.14 there is a problem whereby if someone does a *8 pickup, then sometimes it will keep hold of the channel after seemingly being hung up. The channel shows up if you do core show channels and the device state is IN_USE. I couldn't find a bug report on this - there is https://issues.asterisk.org/view.php?id=18654 but that seems to be different as it blocks all calls whereas in this case everyone else can continue to make and receive calls, just not the person with that sip device. So I was just wondering if anyone had had that experience and if it is known to be fixed in later versions. Thanks, Naomi Rosenberg Developer Data Messaging Communications Ltd t: 0161 850 4005 e: na...@dmcip.com w: www.servicesforasterisk.co.uk -- _ -- 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] Barging in PBX
Hi list, I am using asterisk1.4 pbx , I need to barge of all agents, how can I barge can you help. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 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
[asterisk-users] Know the number of concurrent dial ?
Hi I connected Asterisk 1.6 has several SIP provider, Do you know a tool to make a graph of the number of simultaneous calls incoming and outgoing ? and know the max outgoing call in same time ? thanks Olivier. -- _ -- 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] dundi
Dundi just give you location of extensions. For ring you should have capable dialplan and peering. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali Sent: Wednesday, August 03, 2011 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dundi Dear is it possible to send ring(call) to all devices with same (sip_username) in all servers ? in this schematics, some bodies have shared lines. so all lines must be in service . Best -- Pezhman Lali -- _ -- 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] Changing sip response codes
Say, I've a SIP extension. How can I change the SIP response code to match those needed by the registered SIP device? In this case a Mitel PBX.Tx Shaun -- _ -- 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] trustrpid in sip.conf
Hi Are there any security issues I need to be aware of if I set trustrpid to yes in my sip.conf? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Debugging Sip
Hello, When debugging SIP in Asterisk is it possible to send the SIP debug log to a specific file instead of the general log file, or even better, send each call into its own file for easier analysis? Thanks, Elliot -- _ -- 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 reload, to execute file
Just use a system() call. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Wingrin Sent: Wednesday, August 03, 2011 2:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk reload, to execute file Say, When * reloads it changes the file permissions of below file. How can I call an executable which corrects for this? chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi Tx Shaun -- _ -- 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] Segmentation Fault
On 11-08-03 02:01 AM, Richard Zulu wrote: I have used gdb so that I can perform a backtrace however the program executes and exits without a stack thus not helpful. Any help is appreciated! https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace -- 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
Re: [asterisk-users] Changing sip response codes
On 11-08-03 07:36 AM, Shaun Wingrin wrote: Say, I've a SIP extension. How can I change the SIP response code to match those needed by the registered SIP device? In this case a Mitel PBX.Tx Shaun Why do you need to do this? -- 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
[asterisk-users] Increasing volume ?
Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=10)) SetGlobalVar(Set(VOLUME(RX)=10)) but it does not effect.. any hint ? Zeeshan -- _ -- 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] Increasing volume ?
You need to provide more information - is line in SIP or DAHDI, what release of Asterisk, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Ali Shah Sent: Wednesday, August 03, 2011 9:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Increasing volume ? Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=10)) SetGlobalVar(Set(VOLUME(RX)=10)) but it does not effect.. any hint ? Zeeshan -- _ -- 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 reload, to execute file
On Wed, 3 Aug 2011, Shaun Wingrin wrote: When * reloads it changes the file permissions of below file. Asterisk doesn't change file permissions. Maybe your startup script does? How can I call an executable which corrects for this? You should treat the cause, not the symptom. chmod 777 /var/lib/asterisk/agi-bin/dialparties.agi This is a bad idea. Do you really want anybody to be able to delete or modify this file? What would happen if I were to replace this file with one containing 'rm --force --recursive /' and then call in? '777' says 'I really don't know what I'm doing and I'm too lazy to try to figure it out.' Guess how many files have '777' on any freshly installed OS? -- 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
Re: [asterisk-users] Increasing volume ?
Following :
The Line is SIP,
Asterisk is 1.6.2.5-0ubuntu1.4
Also I tried konference volume [konfernecename] up but it did not increase
*See the extensions.conf where i applied increase volume .
*
-
*SetGlobalVar(Set(VOLUME(TX)=10))
SetGlobalVar(Set(VOLUME(RX)=10)) *
[bigbluebutton]
exten = _.,1,Goto(start-dialplan,s,1)
exten = _.,n,Hangup
[start-dialplan]
exten = s,1,Set(TRIES=1)
exten = s,n,Wait(2)
exten = s,n,Answer
exten = s,n,Goto(prompt,s,1)
[prompt]
exten = s,1,Read(CONF_NUM,conf-getconfno,6,,3,10)
exten = s,n,Goto(bbb-conference,${CONF_NUM},1)
; No need to check if conference is valid as they won't be able to login
; if the conference is invalid.
;
[bbb-voip]
exten = _.,1,Playback(conf-placeintoconf)
; exten = _.,n,MeetMe(${EXTEN},cdMsT)
exten = _.,n,Konference(${EXTEN})
[bbb-conference]
include = echo-test
exten = _.,1,Agi(agi://localhost/findConference?conference=${EXTEN})
exten = _.,n,GotoIf($[${EXTEN} = ${CONFERENCE_FOUND}]?valid:invalid)
exten = _.,n(valid),Playback(conf-placeintoconf)
; exten = _.,n,MeetMe(${CONFERENCE_FOUND},cdMsT)
exten = _.,n,Konference(${CONFERENCE_FOUND})
exten = _.,n(invalid),Goto(handle-invalid-conference,s,1)
[handle-invalid-conference]
exten = s,1,Playback(conf-invalid)
exten = s,n,GotoIf($[${TRIES} 3]?try-again:do-not-try-again)
exten = s,n(try-again),Set(TRIES=$[${TRIES} + 1])
exten = s,n,Goto(prompt,s,1)
exten = s,n(do-not-try-again),Hangup
[echo-test]
;
; Create an extension, 600, for evaluating echo latency.
;
exten = 600,1,Answer ; Do the echo test
exten = 600,n,Playback(demo-echotest) ; Let them know what's going on
exten = 600,n,Echo ; Do the echo test
exten = 600,n,Playback(demo-echodone) ; Let them know it's over
exten = 600,n,Goto(s,6); Start over
-
Zeeshan
On Wed, Aug 3, 2011 at 4:16 PM, Danny Nicholas da...@debsinc.com wrote:
You need to provide more information – is line in SIP or DAHDI, what
release of Asterisk, etc.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Ali Shah
*Sent:* Wednesday, August 03, 2011 9:13 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Increasing volume ?
** **
Hi, I am running asterisk with konference . tried to increase the
conference voice but not success
i tried to add in diaplain
SetGlobalVar(Set(VOLUME(TX)=10))
SetGlobalVar(Set(VOLUME(RX)=10))
but it does not effect..
any hint ?
Zeeshan
--
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Re: [asterisk-users] Increasing volume ?
Hi, In CLI please press Konference then Tab from keyboard then you will see all the command of Konference. You may use AMI connection for batter usw. On 3 Aug 2011 19:46, Danny Nicholas da...@debsinc.com wrote: You need to provide more information - is line in SIP or DAHDI, what release of Asterisk, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Ali Shah Sent: Wednesday, August 03, 2011 9:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Increasing volume ? Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=10)) SetGlobalVar(Set(VOLUME(RX)=10)) but it does not effect.. any hint ? Zeeshan -- _ -- 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] Increasing volume ?
I tried few times this, but no improvement . meeting*CLI konference volume 72193 up any way to do it form extensions.conf in dialplan ? since konfernece name is dynamic Zeeshan On Wed, Aug 3, 2011 at 4:30 PM, virendra bhati virbh...@gmail.com wrote: Hi, In CLI please press Konference then Tab from keyboard then you will see all the command of Konference. You may use AMI connection for batter usw. On 3 Aug 2011 19:46, Danny Nicholas da...@debsinc.com wrote: You need to provide more information - is line in SIP or DAHDI, what release of Asterisk, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Ali Shah Sent: Wednesday, August 03, 2011 9:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Increasing volume ? Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=10)) SetGlobalVar(Set(VOLUME(RX)=10)) but it does not effect.. any hint ? Zeeshan -- _ -- 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] Codec negotiation issue (no audio format found to offer)
From looking into this, it appears as if this is due to Asterisk negotiating the legs separately as if they were not related to the same call. So the ingress leg negotiates ulaw, and despite it knowing that the peer also supports g729 fails the call since it's already decided on ulaw and the egress leg only accepts g729. If this is design intent I'm wondering if there is demand enough to justify a feature request? Any advice on how I can work around this issue? Thanks, -Ryan On Tue, Aug 2, 2011 at 3:51 PM, Ryan McGuire rdmcguir...@gmail.com wrote: Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound to my SIP trunk. Here are the basics of my config, showing the codec list from sip show peer peer: Polycom SP501 (desk phone): -- disallow=all allow=ulawg729 Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) Grandstream HT503 (fxo gateway): -- disallow=all allow=ulawg729 Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) CallWithUs (SIP trunk): -- disallow=all allow=g729 Codecs : 0x100 (g729) Codec Order : (g729:20) When I place an outbound call from the Polycom to callwithus, the invite from the pcom shows both ulaw and g729 in the SDP: INVITE sip:@192.168.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.201;branch=z9hG4bKc8aa981a8B8FF58D From: Office sip:2001@192.168.0.1;tag=4CD2B2A0-B94A2531 To: sip:919785013620@192.168.0.1;user=phone [...] m=audio 2258 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Asterisk sees this: [Aug 2 15:00:31] VERBOSE[1918] chan_sip.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729) The call goes out the callwithus trunk: [Aug 2 15:00:31] VERBOSE[1315] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial(SIP/2001-0047, SIP/CallWithUs/**,300,tTwW) in new stack And then this, no INVITE goes out to callwithus at all: [Aug 2 15:00:31] WARNING[1315] chan_sip.c: No audio format found to offer. Cancelling call to ** [Aug 2 15:00:31] VERBOSE[1315] app_dial.c: -- Couldn't call SIP/CallWithUs/** Similarly, if I set the Grandstream FXO trunk to ulaw only, the call fails as well. It seems as if allowing only a single codec is the issue, if I change the priorities of all codecs to g729 first and ulaw second, the call goes through as g729 successfully. Smells like a bug to me, but I may be overlooking something in my config. Thanks, -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
[asterisk-users] Need a volunteer for a Patch
I tried te route of using iptables and at top production time, it eats 5% of my server, brining it to 95+ CPU usage. Clearly, not an option. I need a patch for chan_sip that when alwaysauthreject=yes does not respond to any REGISTER packet if the username does not exists. I hope that Digium would include this otr similar option in the source code. Alternatively, a new option can be created in sip.conf. I am offering no money for this patch. I think all the community needs this to survive the attack of the evil men from shadowlands. Another nice patch that I already wrote partially, is for cdr_addons_mysql, but it should be included in all cdr-collecting technologies. I just do not save to the database any call that is not connected. This is NOT the same as setting the option at the cdr.conf level. Each cdr technology needs this option as well. I need to save all calls to my cdr_odbc, for ASR calculations, but it is useless to store un-connected calls to mysql, because I use it only as a backup cdr, in case my external SQL Server blows up or has a problem, which happens often. What I did was to hard code this option in the source code, but not including any checkin for a cdr_sql.conf, since I am not a C programmer. -- _ -- 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] snom and srtp
On 03/08/11 03:15, James Perkins wrote: Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). What firmware version have you got on the snom phones? It needs a pretty new version to work properly. I wrote some notes when I got this working here: http://blog.provu.co.uk/item/212/catid/3 Although that was back on Asterisk 1.8.4.1. The same server is currently on 1.8.4.3 and still working OK. cheers, Paul. -- _ -- 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] Increasing volume ?
any way to increase volume in dialplan ? i attached the extensions.conf in privious post Zeeshan On Wed, Aug 3, 2011 at 4:33 PM, Zeeshan Ali Shah zees...@infoshield.infowrote: I tried few times this, but no improvement . meeting*CLI konference volume 72193 up any way to do it form extensions.conf in dialplan ? since konfernece name is dynamic Zeeshan On Wed, Aug 3, 2011 at 4:30 PM, virendra bhati virbh...@gmail.com wrote: Hi, In CLI please press Konference then Tab from keyboard then you will see all the command of Konference. You may use AMI connection for batter usw. On 3 Aug 2011 19:46, Danny Nicholas da...@debsinc.com wrote: You need to provide more information - is line in SIP or DAHDI, what release of Asterisk, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Ali Shah Sent: Wednesday, August 03, 2011 9:13 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Increasing volume ? Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=10)) SetGlobalVar(Set(VOLUME(RX)=10)) but it does not effect.. any hint ? Zeeshan -- _ -- 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] Need a volunteer for a Patch
On 08/03/11 09:49, Venefax wrote: I tried te route of using iptables and at top production time, it eats 5% of my server, brining it to 95+ CPU usage. Clearly, not an option. I need a patch for chan_sip that when alwaysauthreject=yes does not respond to any REGISTER packet if the username does not exists. I hope that Digium would include this otr similar option in the source code. Alternatively, a new option can be created in sip.conf. I am offering no money for this patch. I think all the community needs this to survive the attack of the evil men from shadowlands. Another nice patch that I already wrote partially, is for cdr_addons_mysql, but it should be included in all cdr-collecting technologies. I just do not save to the database any call that is not connected. This is NOT the same as setting the option at the cdr.conf level. Each cdr technology needs this option as well. I need to save all calls to my cdr_odbc, for ASR calculations, but it is useless to store un-connected calls to mysql, because I use it only as a backup cdr, in case my external SQL Server blows up or has a problem, which happens often. What I did was to hard code this option in the source code, but not including any checkin for a cdr_sql.conf, since I am not a C programmer. With your option turned on, evil ones will again be able to enumerate valid usernames. To keep them guessing, you give them the same answer if the user name does not exist or if they gave you a bad password. But with your option turned on, they will know if they have a valid user name or not. Lyle Giese LCR Computer Services, Inc. -- _ -- 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] Need a volunteer for a Patch
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, August 03, 2011 8:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Need a volunteer for a Patch On 08/03/11 09:49, Venefax wrote: I tried te route of using iptables and at top production time, it eats 5% of my server, brining it to 95+ CPU usage. Clearly, not an option. I need a patch for chan_sip that when alwaysauthreject=yes does not respond to any REGISTER packet if the username does not exists. I hope that Digium would include this otr similar option in the source code. Alternatively, a new option can be created in sip.conf. I am offering no money for this patch. I think all the community needs this to survive the attack of the evil men from shadowlands. Another nice patch that I already wrote partially, is for cdr_addons_mysql, but it should be included in all cdr-collecting technologies. I just do not save to the database any call that is not connected. This is NOT the same as setting the option at the cdr.conf level. Each cdr technology needs this option as well. I need to save all calls to my cdr_odbc, for ASR calculations, but it is useless to store un-connected calls to mysql, because I use it only as a backup cdr, in case my external SQL Server blows up or has a problem, which happens often. What I did was to hard code this option in the source code, but not including any checkin for a cdr_sql.conf, since I am not a C programmer. With your option turned on, evil ones will again be able to enumerate valid usernames. To keep them guessing, you give them the same answer if the user name does not exist or if they gave you a bad password. But with your option turned on, they will know if they have a valid user name or not. In SIP the password is not sent until the 2nd packet when authenticating. So even with that patch, you will still respond to the first packet of all register requests for all valid usernames.Not responding in any way to register (and other authentication) requests will help only until the people hacking servers realize what is happening and adapt. -- _ -- 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] Increasing volume ?
On 4/08/11 2:12 AM, Zeeshan Ali Shah wrote: Hi, I am running asterisk with konference . tried to increase the conference voice but not success i tried to add in diaplain SetGlobalVar(Set(VOLUME(TX)=10)) SetGlobalVar(Set(VOLUME(RX)=10)) Should be: SetGlobalVar(VOLUME(TX)=10) SetGlobalVar(VOLUME(RX)=10) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] UK BT ISDN30 settings?
Hello, I'm trying to connect a Digium Wildcard TE110P T1/E1 Card to a 10 channel BT ISDN 30 box, however I'm fairly new to Asterisk and this is the first time I've set-up a telephony interface card can anyone give me some pointers or known working configurations? I'm running Asterisk 1.6 and Dahdi 2.4.1.1. Thanks, Tanuki -- _ -- 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] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?
Can you please elaborate on how to apply the patch? Also, is the repository updated with the new code? Regards, On Tue, Aug 2, 2011 at 7:34 PM, Richard Mudgett rmudg...@digium.com wrote: Can you please point me to the patch that you just made? The patch is committed to v1.6.2 SVN branch. Patch for v1.6.2 only. r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines Asterisk 18103 - Fix reload crash caused by destroying default parking lot Default parking lot was being destroyed in reload and was not being rebuilt properly. This patch keeps features.c reload from destroying the default parking lot in 1.6.2. Bug was caused by a hasty backport which didn't test reload enough times to catch the problem. (closes issue ASTERISK-18103) Reported by: 808blogger Review: https://reviewboard.asterisk.org/r/1337/ Also -r330505 to fix a ref leak with the above patch. Richard -- _ -- 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] TE410P hardware problems
I opened the jumpers on the card putting them in T1 mode and it worked. I had them set to T1 using the options in the dahdi.conf file under /etc/modprobe.d/ That worked well for over a year until it started acting up. Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Tuesday, August 02, 2011 11:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE410P hardware problems If it doesn't go green when you put a hard loopback on the port, then contact Digium support. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Dave George Sent: Tuesday, August 02, 2011 10:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] TE410P hardware problems dmesg: wct4xxp :0a:03.0: SPAN 9: Primary Sync Source wct4xxp :0a:03.0: Span 2 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 10: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 wct4xxp :0a:03.0: Span 3 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 11: Primary Sync Source wct4xxp :0a:03.0: Span 4 configured for ESF/B8ZS wct4xxp :0a:03.0: SPAN 12: Primary Sync Source wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from wct4xxp :0a:03.0: RCLK source set to span 1 -- _ -- 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
