[asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have allowed only one codec(alaw). [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) I also tried to yes/no option transcode_via_sln in asterisk.conf without any success. Any idea? Thanks, --AM -- _ -- 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] Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Hello! In your sip.conf use alaw as your first codec option and see what happens.Best regards, Fellipe Paes Date: Tue, 28 Jun 2011 15:29:11 +0530 From: theasterisk...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asked to transmit frame type slin,while native formats is 0x8 (alaw) Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have allowed only one codec(alaw). [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) I also tried to yes/no option transcode_via_sln in asterisk.conf without any success. Any idea? Thanks, --AM -- _ -- 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] No audio after a reinvite changing codec ---- with SIP DEBUG!!
On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore lmo...@starwon.com.au wrote: On 18/06/2011 5:36 AM, Matteo Campana wrote: Inviato da iPhone Il giorno 16/giu/2011, alle ore 16:37, Eric Wielingewiel...@nyigc.com ha scritto: We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream. Hi Eric, this behavior is an asterisk bug or asterisk can never change the codec on the fly? Thanks, Matteo The problem reported occurs after a fax tone is detected, one might expect T.38 or G711 to be used to handle the fax, not G729! There is no configuration file information for each of the nodes/peers, no debug of each peer involved nor a trace of the packets hence no one will have ideas! Larry. Hi Larry, I have the SIP debug taken from asterisk. In this debug: 1.2.3.4 --- IP SIP PROXY 5.6.7.8 --- IP UAC (Linksys SPA 962) 9.10.11.12 --- IP ASTERISK to connect to the provider 13.14.15.16 -- IP PROVIDER 17.18.19.20 -- IP ASTERISK The SIP debug is available at this link: http://pastebin.com/9DrFDmeC Thanks in advance, Matteo -- __**__**_ -- 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-**usershttp://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] Re connecting to SIP Provider with virtual IP, from pacemaker cluster
Hello Cédric # when the virtual ip come up ip r a SIP_PROVIDER_IP via GATEWAY_IP dev eth0:0 # when the virtual ip come down, maybe facultative because the route is deleted when the interface fall down ip r d SIP_PROVIDER_IP via GATEWAY_IP dev eth0:0 thank's for your hint. i tried it today but connection always come from eth0 and not from the alias eth0:0 and route -n also shows for Iface eth0 and not eth0:0 BR/Torsten -- _ -- 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] Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Thanks for the response. I have disallow=all and allow=alaw in sip.conf for my SIP user. Any other idea? --AM On Tue, Jun 28, 2011 at 4:23 PM, Fellipe Paes fellipe...@hotmail.comwrote: Hello! In your sip.conf use alaw as your first codec option and see what happens. Best regards, Fellipe Paes -- Date: Tue, 28 Jun 2011 15:29:11 +0530 From: theasterisk...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw) Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have allowed only one codec(alaw). [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) I also tried to yes/no option transcode_via_sln in asterisk.conf without any success. Any idea? Thanks, --AM -- _ -- 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] MixMonitor - garbled/corrupted WAV files
Hi, I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The conversation themselves are reportedly of good quality, only the recording is a problem. Hint: this server does not have PRI Digium cards installed (pure SIP, only a transcoder card), while the others do. Could this be the cause? Dahdi_test shows 99.9xxx% accuracy with the dummy timer. Regards, Mike -- _ -- 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] MixMonitor - garbled/corrupted WAV files
On Tue, Jun 28, 2011 at 10:30 AM, Mike l...@net-wall.com wrote: Hi, ** ** I’ve had problems with MixMonitor recordings. A lot (I’d say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. ** ** What can be the cause? The conversation themselves are reportedly of good quality, only the recording is a problem. ** ** Hint: this server does not have PRI Digium cards installed (pure SIP, only a transcoder card), while the others do. Could this be the cause? Dahdi_test shows 99.9xxx% accuracy with the dummy timer. ** ** Regards, ** ** ** ** Mike ** ** I had this happen on a client's server where the HDD was failingmaybe integrity check the disk? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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
Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files
On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote: I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The conversation themselves are reportedly of good quality, only the recording is a problem. Hint: this server does not have PRI Digium cards installed (pure SIP, only a transcoder card), while the others do. Could this be the cause? Dahdi_test shows 99.9xxx% accuracy with the dummy timer. Which version of DAHDI are you using? -- 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
[asterisk-users] Outgoing calls get dropped on high-latency connections.
We're a VoIP provider essentially competing with our local incumbent Telco, and a sizeable number of our customers use satellite internet. As a result, these customers never have ping times less than 500ms, and are often exceeding 2500ms. I manually apply a patch to the Asterisk source code every time we upgrade Asterisk, described here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html This change allows our satellite customers to maintain their SIP connection for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and this version seems to have one very strange bug on these high latency connections. On outgoing and *only* outgoing calls, the call drops after two or three minutes. Incoming calls do not have this problem, so I don't think it's the SIP connection getting killed due to a slow INVITE response. Has anyone heard of this bug? Or should I submit a new bug report to the Asterisk project? This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] Set a specific BLF key on Polycom 650
Hi, I've got a Polycom SPIP 650 3.3.1F-enabled phone with an 14-keys side-car. What is the best way to assign a specific key to a given BLF (without having to assign every previous key) ? At the moment, I'm using settings like this : ... attendant attendant.reg=1 attendant.resourceList.1.address=253 attendant.resourceList.1.label=253 attendant.resourceList.3.address=sip:255 attendant.resourceList.3.label=255 / feature feature.12.name=directed-call-pickup feature.12.enabled=1 / call call.directedCallPickupMethod=legacy call.directedCallPickupString=*8 / It does work but, but in the example above, 253 is assigned to key number 2 and 255 is assigned to key number 3 where key number is the top-most key from phone main body (not from attendant console). 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
Re: [asterisk-users] Outgoing calls get dropped on high-latency connections.
Have you tried SIP session timer values in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar Sent: Tuesday, June 28, 2011 9:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing calls get dropped on high-latency connections. We're a VoIP provider essentially competing with our local incumbent Telco, and a sizeable number of our customers use satellite internet. As a result, these customers never have ping times less than 500ms, and are often exceeding 2500ms. I manually apply a patch to the Asterisk source code every time we upgrade Asterisk, described here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html This change allows our satellite customers to maintain their SIP connection for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and this version seems to have one very strange bug on these high latency connections. On outgoing and *only* outgoing calls, the call drops after two or three minutes. Incoming calls do not have this problem, so I don't think it's the SIP connection getting killed due to a slow INVITE response. Has anyone heard of this bug? Or should I submit a new bug report to the Asterisk project? This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] Using Dial() on SIP and DAHDI connections simultaneously
I think there is a bug in the Dial() application in Asterisk 1.6.2.17 that wasn't present in 1.4.23.1, and I'd like to see if anyone else has this problem. I've been able to reproduce this error: When you use the Dial() command to send a call to both a SIP connection and a DAHDI connection, if the DAHDI connection is busy, the call always gets rejected with an Asterisk message saying the line is busy, even if the SIP connection is not busy. The inverse is not true: if the SIP connection is busy but the DAHDI connection is not, the call goes through to the DAHDI connection without a problem. This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] Polycom BLF
2011/6/20 Gord Urquhart gord...@gmail.com I missed one important parameter in my setup of BLF for polycom phones (at least if you want to do one touch directed pickup) In sip.conf add notifycid=yes the notifycid=yes causes asterisk to add a target uri = callID to the XML of the SIP notify. Without this target uri the Polycom phone will not do a directed pickup. With asterisk 1.6.1.18, I could make this work without setting notifycid=yes isn sip.conf. Hope this helps -- _ -- 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] Outgoing calls get dropped on high-latency connections.
Yes, these are our session-timer settings in sip.conf: session-timers=originate session-expires=600 session-minse=90 session-refresher=uas Quoting Faisal Hanif fai...@vopium.com: Have you tried SIP session timer values in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar Sent: Tuesday, June 28, 2011 9:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing calls get dropped on high-latency connections. We're a VoIP provider essentially competing with our local incumbent Telco, and a sizeable number of our customers use satellite internet. As a result, these customers never have ping times less than 500ms, and are often exceeding 2500ms. I manually apply a patch to the Asterisk source code every time we upgrade Asterisk, described here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html This change allows our satellite customers to maintain their SIP connection for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and this version seems to have one very strange bug on these high latency connections. On outgoing and *only* outgoing calls, the call drops after two or three minutes. Incoming calls do not have this problem, so I don't think it's the SIP connection getting killed due to a slow INVITE response. Has anyone heard of this bug? Or should I submit a new bug report to the Asterisk project? This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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 This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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] MixMonitor - garbled/corrupted WAV files
On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote: I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The conversation themselves are reportedly of good quality, only the recording is a problem. Hint: this server does not have PRI Digium cards installed (pure SIP, only a transcoder card), while the others do. Could this be the cause? Dahdi_test shows 99.9xxx% accuracy with the dummy timer. Which version of DAHDI are you using? 2.4.0 on the problem one. I don't have Dahdi cards though. Mike -- _ -- 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] MixMonitor - garbled/corrupted WAV files
I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The conversation themselves are reportedly of good quality, only the recording is a problem. I had this happen on a client's server where the HDD was failingmaybe integrity check the disk? I will, but voicemails record all right. I will check the HDD integrity just to make sure but this hints to HDD not being the issue. Mike -- _ -- 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 1.6 Dahdi on Centos 5.2
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() /etc/asterisk/sip.conf [general] port = 5060 context = others [2000] type=friend context=my-phones secret=1234 host=dynamic [2001] type=friend context=my-phones secret=1234 host=dynamic [1001] type=friend context=my-phones secret=1234 [] type=friend context=my-phones secret=1234 [phonesys] type=friend username=user1 secret=1234 host=dynamic context=my-phones Any suggestions are welcome. Thanks, motty -- _ -- 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] Set a specific BLF key on Polycom 650
Pretty sure with Polycom's you can only specify in sequential order, you can't pick and choose the buttons you assign. Thanks, --Warren Selby, dCAP On Jun 28, 2011, at 11:58 AM, Olivier oza_4...@yahoo.fr wrote: Hi, I've got a Polycom SPIP 650 3.3.1F-enabled phone with an 14-keys side-car. What is the best way to assign a specific key to a given BLF (without having to assign every previous key) ? At the moment, I'm using settings like this : ... attendant attendant.reg=1 attendant.resourceList.1.address=253 attendant.resourceList.1.label=253 attendant.resourceList.3.address=sip:255 attendant.resourceList.3.label=255 / feature feature.12.name=directed-call-pickup feature.12.enabled=1 / call call.directedCallPickupMethod=legacy call.directedCallPickupString=*8 / It does work but, but in the example above, 253 is assigned to key number 2 and 255 is assigned to key number 3 where key number is the top-most key from phone main body (not from attendant console). 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] MixMonitor - garbled/corrupted WAV files
On Tue, Jun 28, 2011 at 01:24:43PM -0400, Mike wrote: On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote: I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The conversation themselves are reportedly of good quality, only the recording is a problem. Hint: this server does not have PRI Digium cards installed (pure SIP, only a transcoder card), while the others do. Could this be the cause? Dahdi_test shows 99.9xxx% accuracy with the dummy timer. Which version of DAHDI are you using? 2.4.0 on the problem one. I don't have Dahdi cards though. Is the server able to keep acurate time or is there clock drift and/or large skews with ntpd? The timer in the core of DAHDI uses the wall-clock of the system to determine the timing. If that jumps around...I could imagine how you might get garbled recordings (or garbled audio on calls that are in coferences). -- 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
Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() The context in chan_dahdi.conf is my-phones which differs from the my-phone context in extensions.conf. 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
Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
A couple things - First, in extensions.con your context is [my-phone], but you're using my-phones in your dahdi and sip.conf files. Second, you need an 's' extension somewhere in your receiving context in order for asterisk to answer the incoming analog call. Third, I think you've got some issues with your Dial statements, but I'm on my phone right now and can't really diagnose them. I'll take a look later when I'm back at a desk, if no one else has commented by then. Thanks, --Warren Selby, dCAP On Jun 28, 2011, at 12:30 PM, motty.cruz motty.c...@gmail.com wrote: Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() /etc/asterisk/sip.conf [general] port = 5060 context = others [2000] type=friend context=my-phones secret=1234 host=dynamic [2001] type=friend context=my-phones secret=1234 host=dynamic [1001] type=friend context=my-phones secret=1234 [] type=friend context=my-phones secret=1234 [phonesys] type=friend username=user1 secret=1234 host=dynamic context=my-phones Any suggestions are welcome. Thanks, motty -- _ -- 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 1.6 Dahdi on Centos 5.2
My mistake I had fix that typo but no luck Thanks, motty -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Tuesday, June 28, 2011 10:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2 Hello, I have Asterisk 1.6 running on Centos, Also I have one analog telephone line coming on Wildcard TDM400P REV E/F Board 5 I can't get asterisk to dectect call coming from analog line. Here is my /etc/dahdi/system.conf fxsks=1 # global data loadzone = us defaultzone = us /etc/asterisk/chan_dahdi.conf [channels] language=en context=my-phones switchtype=national signalling=fxs_ks channel = 1 /etc/asterisk/extensions.conf [globals] CONSOLE=DAHDI/1 TRUNK=DAHDI/4 TRUNKMSD=1 [my-phone] exten = 2000,1,Dial(DAHDI/1/116) exten = 2000,2,cONGESTION exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline exten = 2001,2,HangUp() exten = 1001,1,Dial(DAHDI/1/7608514114) exten = 1001,2,HangUp() exten = ,1,Dial(DAHDI/1/7608514114) exten = l111,2,HangUp() The context in chan_dahdi.conf is my-phones which differs from the my-phone context in extensions.conf. 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 - No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1388 / Virus Database: 1516/3731 - Release Date: 06/28/11 -- _ -- 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 1.6 Dahdi on Centos 5.2
Thanks Warren,
I have gone ahead and correct my typo. Also, I created 's' extension as you
suggested.
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERIDNAME})
exten = s,n,Wait(4)
exten = s,n,Playback(tt-easels)
exten = s,n,Voicemail(@vm-test)
exten = s,n,Wait(2)
exten = s,n,Playback(vm-goodbye)
exten = s,n,Wait(2)
exten = s,n,HangUp()
I actually followed this e.i
http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html
because I have the same Digium card tdm4oop four modules although I'm only
using one.
Thanks, in advance.
-motty
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Tuesday, June 28, 2011 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2
A couple things -
First, in extensions.con your context is [my-phone], but you're using
my-phones in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in
order for asterisk to answer the incoming analog call.
Third, I think you've got some issues with your Dial statements, but I'm on
my phone right now and can't really diagnose them. I'll take a look later
when I'm back at a desk, if no one else has commented by then.
Thanks,
--Warren Selby, dCAP
On Jun 28, 2011, at 12:30 PM, motty.cruz motty.c...@gmail.com wrote:
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
/etc/asterisk/chan_dahdi.conf
[channels]
language=en
context=my-phones
switchtype=national
signalling=fxs_ks
channel = 1
/etc/asterisk/extensions.conf
[globals]
CONSOLE=DAHDI/1
TRUNK=DAHDI/4
TRUNKMSD=1
[my-phone]
exten = 2000,1,Dial(DAHDI/1/116)
exten = 2000,2,cONGESTION
exten = 2001,1,Dial(DAHDI/1/7608514114) ; analog phoneline
exten = 2001,2,HangUp()
exten = 1001,1,Dial(DAHDI/1/7608514114)
exten = 1001,2,HangUp()
exten = ,1,Dial(DAHDI/1/7608514114)
exten = l111,2,HangUp()
/etc/asterisk/sip.conf
[general]
port = 5060
context = others
[2000]
type=friend
context=my-phones
secret=1234
host=dynamic
[2001]
type=friend
context=my-phones
secret=1234
host=dynamic
[1001]
type=friend
context=my-phones
secret=1234
[]
type=friend
context=my-phones
secret=1234
[phonesys]
type=friend
username=user1
secret=1234
host=dynamic
context=my-phones
Any suggestions are welcome.
Thanks,
motty
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-
No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1388 / Virus Database: 1516/3731 - Release Date: 06/28/11
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Re: [asterisk-users] t.38 virtual fax software?
What you say...David Backeberg (dbackeb...@gmail.com): On Fri, Jun 24, 2011 at 4:55 PM, Hose hose+aster...@bluemaggottowel.com wrote: Can anyone recommend some kind of virtual t.38 fax software? I'd like to test/debug some of the t.38 stuff, but it'd be much easier if I had a software client that could just generate the faxes from a workstation, rather than having to sit with the fax machine + t.38 ata to source faxes from. another machine or virt + asterisk + callfiles that invoke SendFax() There doesn't seem to be much out there, and the stuff that's out there is kind of expensive for me just to be using for testing. That's because callfiles are free Hmmm, I suppose that makes sense with the limited amount of fax-stack software options out there. Guess I'll be rolling out a new VM. hose -- _ -- 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] Add # at the end of dialled number
Hi all,
Anybody know if is it possible to add # at the end of dialled number ?
kinda : exten = _00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4},25,T)
In this line I am switching the C.O code but , how could I put # automatic at
the end ?
Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda --
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Re: [asterisk-users] Add # at the end of dialled number
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Tuesday, June 28, 2011 1:22 PM
To: Asterisk Asterisk
Subject: [asterisk-users] Add # at the end of dialled number
Hi all,
Anybody know if is it possible to add # at the end of dialled number ?
kinda : exten = _00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4},25,T)
In this line I am switching the C.O code but , how could I put # automatic
at the end ?
Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Have you tried this?
kinda : exten = _00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4}#,25,T)
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Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, June 28, 2011 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MixMonitor - garbled/corrupted WAV files On Tue, Jun 28, 2011 at 01:24:43PM -0400, Mike wrote: On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote: I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of those are corrupted (can`t be opened) or garbled. That is on only one server, which is using the same Asterisk version (1.6.2.18) as the other servers which are mostly fine. What can be the cause? The conversation themselves are reportedly of good quality, only the recording is a problem. Hint: this server does not have PRI Digium cards installed (pure SIP, only a transcoder card), while the others do. Could this be the cause? Dahdi_test shows 99.9xxx% accuracy with the dummy timer. Which version of DAHDI are you using? 2.4.0 on the problem one. I don't have Dahdi cards though. Is the server able to keep acurate time or is there clock drift and/or large skews with ntpd? The timer in the core of DAHDI uses the wall-clock of the system to determine the timing. If that jumps around...I could imagine how you might get garbled recordings (or garbled audio on calls that are in coferences). I understand garbled sounds being related to timing source (Will check it) but often the file simply cannot be opened by any media player. As if the format was simply corrupted. This isn't due to timing, I would guess. Mike -- _ -- 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] Set a specific BLF key on Polycom 650 [SOLVED]
2011/6/28 Warren Selby wcse...@selbytech.com Pretty sure with Polycom's you can only specify in sequential order, you can't pick and choose the buttons you assign. Thanks, --Warren Selby, dCAP On Jun 28, 2011, at 11:58 AM, Olivier oza_4...@yahoo.fr wrote: Hi, I've got a Polycom SPIP 650 3.3.1F-enabled phone with an 14-keys side-car. What is the best way to assign a specific key to a given BLF (without having to assign every previous key) ? At the moment, I'm using settings like this : ... attendant attendant.reg=1 attendant.resourceList.1.address=253 attendant.resourceList.1.label=253 attendant.resourceList.3.address=sip:255 attendant.resourceList.3.label=255 / feature feature.12.name=directed-call-pickup feature.12.enabled=1 / call call.directedCallPickupMethod=legacy call.directedCallPickupString=*8 / It does work but, but in the example above, 253 is assigned to key number 2 and 255 is assigned to key number 3 where key number is the top-most key from phone main body (not from attendant console). 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 That's the answer I feared ... Thanks for helping, 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
[asterisk-users] IVR
I need to make an IVR as follows:
1 an incoming call and run an
AGI script to alert the database, everything perfect here.
2 Play a
music on hold and executes a loop while searching the database for a
change in a field when the field change, cut the music on hold and keep
doing things. I can do this?
I solved it by doing the following:
exten
= 4321,n,Answer()
exten = 4321,n,AGI(script.agi,${UNIQUEID},WAITING)
;Plays music with duration of 5 seconds
exten =
4321,n,Playback(waiting-bucle-audio)
exten =
4321,n,AGI(script-test.agi,${UNIQUEID})
;If script return variable
${state} with text TRANSFERING goto call
exten =
4321,n,GotoIf($[${state} = TRANSFERING]?call)
exten =
4321,n,Goto(context,4321,3)
exten = 4321,n(call),Dial(SIP/${sip},,20)
exten = 4321,n,Hangup()
anyone have any idea how I can run a script
in a loop until I return what I want and cut the music on hold?
any
help is welcome
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Re: [asterisk-users] test call generator
Hi List, I'm trying to get working SIPp with media but something is wrong (it's working well without media), please help: This is the command I send at SIPp server: ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err This is the result I see: Last Error: Aborting call on unexpected message for Call-Id '19-12768@12... What I see at logs: 2011-06-28 14:32:57:6241309289577.624809: Aborting call on unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' (index 1), received 'SIP/2.0 488 Not acceptable here^M Via: SIP/2.0/UDP 127.0.0.1:5061 ;branch=z9hG4bK-12768-1-0;received=192.168.25.253^M From: sipp sip:sipp@127.0.0.1:5061;tag=12768SIPpTag091^M To: sut sip:2005@192.168.1.18:5060;tag=as3614adc3^M Call-ID: 1-12768@127.0.0.1^M CSeq: 1 INVITE^M Server: Asterisk PBX 1.8.4.1^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH^M Supported: replaces, timer^M Content-Length: 0^M This is my asterisk 1.8's configuration: *sip.conf* [sipp] type=friend context=sipp host=dynamic port=6000 user=sipp canreinvite=no disallow=all allow=ulaw * * *extensions.conf:* [sipp] exten = 2005,1,Answer same=n,Dial(SIP/intern,30) same=n,Hangup() exten = 2006,1,Answer() same= n,WaitMusicOnHold(4) same= n,Hangup() I'm using sipp.3.1.src.tar.gz and I have installed it this way: ..sip.svn# make pcapplay Thanks in advance. Elder On Thu, May 12, 2011 at 2:51 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: http://tinyurl.com/3hx5652 On Thu, May 12, 2011 at 11:52 AM, Daniel - Asterisk earohua...@gmail.comwrote: Hello Everyone, I wonder if someone could share a manual about using SIPp for Asterisk's testing. I'll be gratefull Regards, Elder Arohuanca Lima - Peru On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com wrote: Sipp looks pretty good! I don't know how I missed this one. This would've saved me tons of time a couple months ago. I plan on using it to load test using 2 Asterisk servers, one to initiate the SIP calls, the other to receive. Thanks for the tip Alex. Zac Wolfe Safi Systems LLC www.safisystems.com On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov abalas...@evaristesys.com wrote: What you are looking for is SIPP: http://sipp.sourceforge.net/ It won't intrinsically tell you anything about the data; it's up to you to appropriate the findings. But it accomplishes the generation of traffic (and dummy media!) on a technical level. Igor Hernandez wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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] IVR
Waitexten is preferable to using the AGI method
Exten = 4321,n,waitexten(5) - just wait 5 seconds
Exten = 4321,n,waitexten(5,m) – wait with music
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ezequiel Lovelle
Sent: Tuesday, June 28, 2011 2:35 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] IVR
I need to make an IVR as follows:
1 an incoming call and run an AGI script to alert the database, everything
perfect here.
2 Play a music on hold and executes a loop while searching the database for a
change in a field when the field change, cut the music on hold and keep doing
things. I can do this?
I solved it by doing the following:
exten = 4321,n,Answer()
exten = 4321,n,AGI(script.agi,${UNIQUEID},WAITING)
;Plays music with duration of 5 seconds
exten = 4321,n,Playback(waiting-bucle-audio)
exten = 4321,n,AGI(script-test.agi,${UNIQUEID})
;If script return variable ${state} with text “TRANSFERING” goto “call”
exten = 4321,n,GotoIf($[${state} = TRANSFERING]?call)
exten = 4321,n,Goto(context,4321,3)
exten = 4321,n(call),Dial(SIP/${sip},,20)
exten = 4321,n,Hangup()
anyone have any idea how I can run a script in a loop until I return what I
want and cut the music on hold?
any help is welcome
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[asterisk-users] AST-2011-011: Possible enumeration of SIP users due to differing authentication responses
Asterisk Project Security Advisory - AST-2011-011 ++ | Product | Asterisk | |+---| | Summary | Possible enumeration of SIP users due to | || differing authentication responses| |+---| | Nature of Advisory | Unauthorized data disclosure | |+---| | Susceptibility | Remote unauthenticated sessions | |+---| | Severity | Moderate | |+---| | Exploits Known | No| |+---| |Reported On | June 11, 2011 | |+---| |Reported By | | |+---| | Posted On | June 28, 2011 | |+---| | Last Updated On | June 28, 2011 | |+---| | Advisory Contact | Terry Wilson twil...@digium.com | |+---| | CVE Name | CVE-2011-2536 | ++ ++ | Description | Asterisk may respond differently to SIP requests from an | | | invalid SIP user than it does to a user configured on| | | the system, even when the alwaysauthreject option is set | | | in the configuration. This can leak information about| | | what SIP users are valid on the Asterisk system. | ++ ++ | Resolution | Respond to SIP requests from invalid and valid SIP users | || in the same way. Asterisk 1.4 and 1.6.2 do not respond| || identically by default due to backward-compatibility | || reasons, and must have alwaysauthreject=yes set in| || sip.conf. Asterisk 1.8 defaults to alwaysauthreject=yes. | || | || IT IS ABSOLUTELY IMPERATIVE that users of Asterisk 1.4| || and 1.6.2 set alwaysauthreject=yes in the general section | || of sip.conf. | ++ ++ | Affected Versions| || | Product | Release Series || |--++| | Asterisk Open Source | 1.4.x | All versions | |--++| | Asterisk Open Source |1.6.2.x | All versions | |--++| | Asterisk Open Source | 1.8.x | All versions | |--++| |Asterisk Business Edition | C.3.x | All versions | ++ ++ | Corrected In | || | Product | Release | |--+-| | Asterisk Open Source |1.4.41.2, 1.6.2.18.2, 1.8.4.4|
[asterisk-users] Clarification of the terms shown on CLI
Hi everyone, When doing a sip show settings on Asterisk 1.6.2.18, I see the following: Match Auth Username:No Allow unknown access: Yes Allow subscriptions:Yes Allow overlap dialing: Yes Allow promsic. redir: No Enable call counters: No What do each of above signify? Thanks -- _ -- 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 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 Now Available (Security Releases)
The Asterisk Development Team has announced the release of Asterisk versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security releases. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the following issue: AST-2011-011: Asterisk may respond differently to SIP requests from an invalid SIP user than it does to a user configured on the system, even when the alwaysauthreject option is set in the configuration. This can leak information about what SIP users are valid on the Asterisk system. For more information about the details of this vulnerability, please read the security advisory AST-2011-011, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.4 Security advisory AST-2011-011 is available at: http://downloads.asterisk.org/pub/security/AST-2011-011.pdf Thank you for your continued support of Asterisk! -- _ -- 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] Clarification of the terms shown on CLI
On Tue, Jun 28, 2011 at 4:53 PM, Bruce B bruceb...@gmail.com wrote: Hi everyone, When doing a sip show settings on Asterisk 1.6.2.18, I see the following: Match Auth Username: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Allow promsic. redir: No Enable call counters: No What do each of above signify? Thanks from http://svn.asterisk.org/svn/asterisk/trunk/configs/sip.conf.sample ;match_auth_username=yes; if available, match user entry using the ; 'username' field from the authentication line ; instead of the From: field. ;allowguest=no ; Allow or reject guest calls (default is yes) ; If your Asterisk is connected to the Internet ; and you have allowguest=yes ; you want to check which services you offer everyone ; out there, by enabling them in the default context (see below). ;allowsubscribe=no ; Disable support for subscriptions. (Default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) --- btw this one is funny ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; Note that promiscredir when redirects are made to the ; local system will cause loops since Asterisk is incapable ; of performing a hairpin call. ;callcounter = yes ; Enable call counters on devices. This can be set per ; device too. -- ~~~ Andrew lathama Latham lath...@gmail.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
Re: [asterisk-users] Add # at the end of dialled number
works !
Thanks!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 28 Jun 2011 13:31:41 -0500
Subject: Re: [asterisk-users] Add # at the end of dialled number
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Tuesday, June 28, 2011 1:22 PM
To: Asterisk Asterisk
Subject: [asterisk-users] Add # at the end of dialled number
Hi all, Anybody know if is it possible to add # at the end of dialled number
? kinda : exten = _00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4},25,T) In
this line I am switching the C.O code but , how could I put # automatic at the
end ? Thanks in advanced!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda Have you tried this?kinda : exten =
_00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4}#,25,T)
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[asterisk-users] for suport server
Dear sir, i would like to request you please support your web . do some work your web , please accept my request. Thanks akram Dhaka, Bangladesh -- _ -- 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
