Re: [asterisk-users] STIR/SHAKEN
Sebastian, There are many reasons why someone would want the DIDs provided by one provider and outbound calls to go out via 1,2 3, or more providers. In one of my installs I have a situation where local calls are placed via a local telco switch but LD calls go out via a voip provider. The Local telco has the DID but the LD does not so I have to verify the DIDs with the Voip provider(s). Another case may be for least cost routing. There are other reasons but you can see that it is not always as simple as using the same provider for DID and origination. Thanks, John On 3/11/21 3:34 PM, Sebastian Nielsen wrote: I reallt don’t understand why people simply use the same operator to terminate your calls, which also provide DIDs for you. Then you don’t need to touch this at all, your carrier will do all the STIR/SHAKEN handling for you, you are just a PBX customer. And then the operator then simply limits your account to only present your DID as outgoing number. Seems to be a unneccesary complicated solution just to have your numbers at company 1 and have your call termination at company 2. So fricking unneccessary. What I know there is requirements of number portability, so as long as company 2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from company 1 to company 2 – then company 2 owns your DIDs. Best regards, Sebastian Nielsen *Från:* asterisk-users-boun...@lists.digium.com *För *Alexander Perkins *Skickat:* den 12 mars 2021 01:23 *Till:* asterisk-users@lists.digium.com *Ämne:* Re: [asterisk-users] STIR/SHAKEN Hi Jeff. What exactly do you mean by the 'inbound piece'? I've spent quite a lot of time with the folks at TILTX understanding the framework; but I am not exactly sure what you mean by the 'inbound piece. Greg/Doug, like many folks here, we use LCR. So, the terminating carrier is not necessarily the one that issued us the telephone numbers. So, they will not sign it or simply cannot sign it. Remember that a very limited number of companies can actually sign the calls; the rest have to buy it from these 'Service Providers'. And there is another situation - the company you purchase your numbers from and the company you place your calls through may be different and both may not be able to sign your calls. Again, a very limited number of service providers that can actually sign your calls. So what do you do in that scenario? You have to find a Service Provider that can: 1. Verify you own that telephone number(s). 2. Sign your calls. 3. Provide you with the technical means to do so. So, that's that... I hope this makes sense. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
On 03/13/2014 01:13 PM, Ron Wheeler wrote: -1 Prefer top posting. Easy to see if I want to scroll down to see if it is something interesting to me. I get a lot of e-mails each day and scrolling wastes too much time. But if you have a solution to a problem that I raise, please feel free to post it anywhere you like. On 13/03/2014 11:33 AM, A J Stiles wrote: (If you want to reply to this message, this is not where your reply goes) Please, for the benefit of anyone reading the archives in search of answers to a question, when replying to messages on this list, can everyone try to follow the natural flow of conversation? That is, position your reply *AFTER* the thing you are replying to, not before it. You may remove quoted material in order to keep the message size down, but please leave enough of it to preserve context. (If you want to reply to a point made in the preceding paragraph, this is where your reply goes) If you need to make a point-by-point argument, split up your reply -- inserting artificial paragraph breaks into the quoted material, if necessary -- so each section of your reply follows the point it is addressing. (If you want to reply to a point made in the preceding paragraph, or the message as a whole, this is where your reply goes) This war comes up often, too often! Good manors are becoming a thing of the past and this is a sad thing in my opinion. It has been my experience that most people on this list prefer the older tried and tested method of bottom posting. Makes it better for people who find the post later while googleing a problem they are having. Question before answers. If you feel that you do not have enough time to take the 1 maybe two seconds to scroll to the bottom of a post, you are way to busty and need to re-prioritize you activities. Obviously this is just my opinion, take or leave it, your choice. Thank You, JohnM -- _ -- 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] Phone - NAT/FIREWALL - Internet - NAT/Firewall- Asterisk
Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the phone to my Cell, my cell rings once and then stops. I get a packet retransmission error: WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical Response) Followed by: n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no reply to our critical packet I am assuming that there is a problem with NAT. I have externip set in sip.conf. Any pointers to what I am missing? Thanks, JohnM -- _ -- 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] Phone - NAT/FIREWALL - Internet - NAT/Firewall- Asterisk
top posting so as to not make thread even more confusing. Nick, I have nat=force_rport,comedia in sip.conf. It is my understanding that nat=yes is deprecated? Thanks, JohnM On 01/02/2014 10:51 AM, Nick Olsen wrote: Make sure you have nat=yes in your sip.conf either under globals or individual sip peer settings. Nick Olsen Network Operations (855) FLSPEED x106 *From*: John Millican j...@millican.us *Sent*: Thursday, January 02, 2014 10:50 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: [asterisk-users] Phone - NAT/FIREWALL - Internet - NAT/Firewall- Asterisk Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the phone to my Cell, my cell rings once and then stops. I get a packet retransmission error: WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical Response) Followed by: n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no reply to our critical packet I am assuming that there is a problem with NAT. I have externip set in sip.conf. Any pointers to what I am missing? Thanks, JohnM -- _ -- 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] Phone - NAT/FIREWALL - Internet - NAT/Firewall- Asterisk
Adam, Thanks, I will try that this afternoon. JohnM On 01/02/2014 11:31 AM, Adam Moffett wrote: top posting is superior anyway --- *ducking to avoid thrown objects* If I recall correctly, when doing something like that with a polycom I had to set the registration interval absurdly low, like 20 seconds or something. I think the Polycom didn't send keepalives and that was the workaround. top posting so as to not make thread even more confusing. Nick, I have nat=force_rport,comedia in sip.conf. It is my understanding that nat=yes is deprecated? Thanks, JohnM On 01/02/2014 10:51 AM, Nick Olsen wrote: Make sure you have nat=yes in your sip.conf either under globals or individual sip peer settings. Nick Olsen Network Operations (855) FLSPEED x106 *From*: John Millican j...@millican.us *Sent*: Thursday, January 02, 2014 10:50 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: [asterisk-users] Phone - NAT/FIREWALL - Internet - NAT/Firewall- Asterisk Hello, CentOS 6.x and Asterisk 11.x I have an interesting, to me at least, situation. Using a Polycom 501(also tried with X-Lite). I have set up Asterisk to accept registration from the Polycom and it registers successfully but then withing 30 seconds on the CLI I get the message that the Polycom is unreachable. The phone still shows that it is registered and if I try to place a call from the phone to my Cell, my cell rings once and then stops. I get a packet retransmission error: WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical Response) Followed by: n_sip.c:4203 retrans_pkt: Hanging up call xx@192.168.0.100 - no reply to our critical packet I am assuming that there is a problem with NAT. I have externip set in sip.conf. Any pointers to what I am missing? Thanks, JohnM -- _ -- 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 on Windows
On 12/04/2013 11:00 AM, Paul Belanger wrote: On 13-12-04 10:19 AM, CDR wrote: Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. Note: I am a Linux expert user, but it took me years to get here. And still, moving from regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET framework and Windows server 2012 are miles away in terms of friendliness and on equal footing on performance. I don´t mean another slow cygwin port, I man a native Asterisk for windows. In fact, I would invest on the project if somebody wants to do it. Do you just sit around and think shit up to blame Digium all day? Normally I do not respond to trolls but... If you want an Asterisk version to run on Windows, go for it. You are free to create it yourself. Most of the folks on this list realize the Asterisk on Windows is a huge mistake. If you really believe that this is such a good idea, go for it and become a bazillionare from your work. Then you can come back and say I told you so. Until then take the advise of the many good folks on this list that collectively have many decades of experience and run asterisk on Linux. Regards, JohnM -- _ -- 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] POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)
Hello,
I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running
Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia,
USA. Whenever I place a call to one of the two lines I get a red alam
and then it clears and repeats this till I hang up. There is no caller
ID on the Line (boss won't pay for it).
Any help is most appreciated.
TIA,
JohnM
lspci relevent output:
08:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
cat /proc/interrupts:
19: 54390 1286431613 IO-APIC-fasteoi wctdm
(no shared interupts)
dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState
pseudodefault
default In Service
1altrurstn
default In Service
2altrurstn
default In Service
PBX1:/home/jmillican# dahdi_cfg -vvv
DAHDI Tools Version - 2.5.0.1
DAHDI Version: 2.5.0.1
Echo Canceller(s): HWEC, MG2
Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
2 channels to configure.
Setting echocan for channel 1 to mg2
Setting echocan for channel 2 to mg2
in chan_dahdi.conf
[channels]
context=altrurstn
signalling=fxs_ks
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
echotraining=800
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
#include dahdi_additional.conf
#include dahdi-channels.conf
in dahdi-channels.conf
; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER)
;;; line=1 WCTDM/4/0
signalling=fxs_ks
callerid=asreceived
group=0
context=altrurstn
channel = 1
callerid=
group=
context=altrurstn
;;; line=2 WCTDM/4/1
signalling=fxs_ks
callerid=asreceived
group=0
context=altrurstn
channel = 2
callerid=
group=
context=altrurstn
/etc/dahdi/modules loads only wctdm.
/etc/dahdi/system.conf:
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
Relevent Extensions.conf:
[altrurstn-in]
exten = s,1,Wait(1);
exten = s,n,Set(CDR(accountcode)=fromoustide)
exten = s,n,Set(CDR(userfield)=POTS-${EXTEN})
exten = s,n,GoTo(999,1);
exten = 999,1,Answer();
exten = 999,n,NoOp(${CALLERID(all)});
exten = 999,n,wait(1);
exten = 999,n,Set(foo=0);
exten = 999,n,Set(count=0);
exten = 999,n,Read(foo,00010002,4,,,2);
exten = 999,n,GoToIf($[${foo}=9]?directory);
exten = 999,n,GoToIf($[${foo}=0]?oper)
exten = 999,n,GoToIf($[${LEN(${foo})} 4]?restart:altrurstn,${foo},1);
exten = 999,n(restart),Set(COUNT=$[${COUNT} + 1]);
exten = 999,n,NoOp(${COUNT});
exten = 999,n,GoToIf($[${COUNT} 1]?oper:continue);
exten = 999,n(continue),Read(foo,0002,4,,,2);
exten = 999,n,GoToIf($[${foo}=9]?directory);
exten = 999,n,GoToIf($[${foo}=0]?oper)
exten = 999,n,GoToIf($[${LEN(${foo})}4]?restart:altrurstn,${foo},1);
exten = 999,n(oper),GoTo(0,1);
exten = 999,n(directory),Directory(default,altrurstn,p(500));
exten = 999,n,Hangup();
What I get in the CLI:
[Apr 26 19:26:53] -- Starting simple switch on 'DAHDI/1-1'
[Apr 26 19:26:53] -- Executing [s@altrurstn-in:1] Wait(DAHDI/1-1,
1) in new stack
[Apr 26 19:26:54] -- Executing [s@altrurstn-in:2] Set(DAHDI/1-1,
CDR(accountcode)=fromoustide) in new stack
[Apr 26 19:26:54] -- Executing [s@altrurstn-in:3] Set(DAHDI/1-1,
CDR(userfield)=POTS-s) in new stack
[Apr 26 19:26:54] -- Executing [s@altrurstn-in:4] Goto(DAHDI/1-1,
999,1) in new stack
[Apr 26 19:26:54] -- Goto (altrurstn-in,999,1)
[Apr 26 19:26:54] -- Executing [999@altrurstn-in:1]
Answer(DAHDI/1-1, ) in new stack
[Apr 26 19:26:54] -- Executing [999@altrurstn-in:2]
NoOp(DAHDI/1-1, ) in new stack
[Apr 26 19:26:54] -- Executing [999@altrurstn-in:3]
Wait(DAHDI/1-1, 1) in new stack
[Apr 26 19:26:55] WARNING[11189]: chan_dahdi.c:7728 handle_alarms:
Detected alarm on channel 1: Red Alarm
[Apr 26 19:26:55] == Spawn extension (altrurstn-in, 999, 3) exited
non-zero on 'DAHDI/1-1'
[Apr 26 19:26:55] -- Hanging up on 'DAHDI/1-1'
[Apr 26 19:26:55] -- Hungup 'DAHDI/1-1'
[Apr 26 19:26:58] NOTICE[11159]: sig_analog.c:3709
analog_handle_init_event: Alarm cleared on channel 1
[Apr 26 19:26:59] -- Starting simple switch on 'DAHDI/1-1'
[Apr 26 19:26:59] -- Executing [s@altrurstn-in:1] Wait(DAHDI/1-1,
1) in new stack
[Apr 26 19:27:00] -- Executing [s@altrurstn-in:2] Set(DAHDI/1-1,
CDR(accountcode)=fromoustide) in new stack
[Apr 26 19:27:00] -- Executing [s@altrurstn-in:3] Set(DAHDI/1-1,
CDR(userfield)=POTS-s) in new stack
[Apr 26 19:27:00] -- Executing [s@altrurstn-in:4] Goto(DAHDI/1-1,
999,1) in new stack
[Apr 26 19:27:00] -- Goto (altrurstn-in,999,1)
[Apr 26 19:27:00] --
[asterisk-users] using AMI and Telnet to place calls
Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 No Circuit Available back from xxx.xxx.xxx.xxx:5060 [Mar 1 12:55:50] Channel SIP/provider was never answered and be able to relate that back to the dialed number for that call. Is this possible? I am using async in the AMI command. Do I need to do something such as adding and event id to the AMI originate action then listen for response from AMI? Obviously I am a bit lost here. Thanks for any pointers toward the solution. JohnM -- _ -- 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] Capture sip Response
Hello, I am using a mix of Call files and AMI telnet from a perl app to place calls. I sometimes get this in the CLI: -- Attempting call on sip/551234@providerfor 1@mycontext:1 (Retry 1) [Feb 27 13:47:07] == Using SIP RTP CoS mark 5 [Feb 27 13:47:07] -- Got SIP response 503 No Circuit Available back from xxx.xxx.xxx.xxx:5060 [Feb 27 13:47:07] Channel SIP/provider was never answered. I would like to be able to capture the Got SIP response 503 No Circuit Available back from xxx.xxx.xxx.xxx:5060 line in a var to be used by a perl AGI that inserts to a mongoDB for reporting. Is this possible? I have read many articles about using hangupcause and siphangupcause but they do not provide the same information I believe because the call was never answered so hangup does not apply. TIA, JohnM -- _ -- 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] A new hack?
On 12/2/2011 12:44 PM, Steve Edwards wrote: On Fri, 2 Dec 2011, Jim Lucas wrote: How is using Fail2Ban less resource intensive then me writing (by hand) iptable rules? It depends on how you define resources and how much of those resources you have. Gordon (based on my understanding of his posts) does a lot of Asterisk systems on very limited hardware hosts. His approach uses iptables features to limit the number of SIP INVITES and REGISTERS per second per IP address. Thus, Gordon's approach is more responsive (since it doesn't require periodic log file scanning) and requires less hardware resources (since it doesn't depend on running relatively 'slothish' resource intensive script interpreters like Perl or PHP periodically). If you have limited admin skills and more hardware resources, F2B makes sense. If you have more admin skills and limited hardware resources, Gordon's approach makes more sense. Personally, I find any approach that tracks log files 'hackish' but if you centralize your logging (which I always do) it does allow you to detect patterns of abuse across multiple hosts. Now this, I would say was very well put. As always, just my opinion. JohnM -- _ -- 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] A new hack?
On 11/29/2011 12:48 PM, C F wrote: On Mon, Nov 28, 2011 at 10:57 AM, Tom Browningttbrown...@gmail.com wrote: On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson gordon+aster...@drogon.net wrote: Linux has excellent built-in subsystems to control firewalling and so on without resorting to external programs. It's called iptables. If you know how to use them, then using an external resource such as fail2ban is unneccessary. That's like saying you don't need FreePBX because you have this thing called Asterisk. Very well put. -- This may well turn out to just be troll fodder but I can not resist. I disagree with the above being very well put, personally I think it is the opposite of well put. Maybe I am misunderstanding the gist of the comment but, I do not NEED FreePBX, I have Asterisk makes perfect sense to me. I have been using asterisk for a few years now and have not yet found anything that I need to do with Asterisk that I must have FreePBX to accomplish. Could I do the same things with FreePBX on top of Asterisk, maybe. I am not an expert in iptables but I have been semi successful in adapting what others have done to fit my needs. I have found this to work better FOR ME than Fail2ban. I have used and will continue to use Fail2ban for other purposes because I am not an iptables expert. In my opinion one should find the tools that work best for you in your situation and use them. You may well change your mind in the future but that is the beauty of this industry, it changes all the time, what I feel works best today may well not be what I think works best tomorrow as new tools are developed and proven and also as I become more experianced with the old tried and true tools. As usual, just my 2 cents (US currency, exchange rates not compensated for) JohnM -- _ -- 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] Recommendations
On 11/28/2011 3:35 PM, Danny Nicholas wrote: If you put a gun to my head I would say to stay with Centos 5 and either 1.4.42 or 10.0.0-rc2. 10.0.0-rc2 removes a feature that was killing me in 1.4, but if you aren't doing IVR stuff, you can stay with what you know. Another thing to consider though; 1.4.8 is prior to the Zaptel-to-Dahdi conversion so that might cause you some joy. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier *Sent:* Monday, November 28, 2011 2:31 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Recommendations I am currently running Asterisk 1.4.8 and have been for quite a while, it has served me well. Getting ready to build a new box to replace the existing installation of Asterisk. My primary use of the Asterisk box is run queues. I am sure the queue features and functionality have been updated, expanded since 1.4.8 and I am wondering what version of Ast you guys would recommend. Looking for the best version in terms of queue features, functionality. Also, an OS recommendation would be great. Been running on CentOS forever and no reason to want to change. Just looking for the best version of CentOS to run the best/stable version of Asterisk on. To be clear: Recommended: Asterisk Version: OS Version: I am even thinking about using AsteriskNow, don't need the FreePBX but I have worked with it before and it used to be possible to still do custom stuff and co-exist with FreePBX. I like having FreePBX available for the simple stuff so it's not a bad thing if it's there. Does it have an intergrated web server that I could run a lightweight control panel on? That would be another plus. Thanks in advance for any help, been out of touch for a while. I will be doing my research and lots of reading over the next few days but thought it couldn't hurt to see what the general consensus is on these topics. Danny, Can you expand on what feature 10.0.0-rc2 removed that was causing you problems with IVR? I am starting to undertake some major IVR scripting so am rather curious. Thanks, JohnM -- _ -- 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] Maybe slightly OT but..
Thanks to all for the responses. Boss calls overseas a lot and has an unlimited data plan, so this coupled with the rates that we get for our VoIP calls it is much cheaper than what Verizon charges. JohnM On 10/11/2011 1:29 AM, Jeremy Kister wrote: On 10/10/2011 10:08 PM, Andres wrote: I would recommend Acrobits. Not free but only a few bucks. It works fine with ATT 3G. +1 only thing i like better is it's big brother, Groundwire -- _ -- 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] Maybe slightly OT but..
Hello all, Does anyone know of a good free/inexpensive 3G SIP client for the iPhone? If anyone is using one that works good for them could you please let me know. Thank You, JohnM -- _ -- 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] AMI Commands - not working as Expected, Maybe???
On Tue, Aug 16, 2011 at 4:42 AM, john Millican j...@millican.us
mailto:j...@millican.us wrote:
On 8/15/2011 5:48 PM, john Millican wrote:
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI. I have
tried all sorts of combinations,entering them after making a
connection to ami through telnet, of the many available
examples on voip-info.org http://voip-info.org such as:
Action: Originate
Channel: sip/xx@xxx
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: test1
Exten: acs1
Priority: 1
CallerID: xx
Account: MyTest
Command: Set(var1=123456)
Command: Set(var2=54321)
also tried:
Var:
Variable:
SetVar:
Each individually for the two variables I need and both on the
same line separated by a | or a ,
Always when I hit return twice to give the \r\n\r\n The call
is successful but where I have
exten = acs1,n,NoOp(Vars = ${var1}, ${var2});
in my dialplan what I get is:
[2011-08-15 17:20:28] -- Executing [acs1@test1:2]
NoOp(SIP/xxx-0451, Vars = | ) in new stack
Obviously not what I was hoping for.
Any help would be greatly appreciated.
TIA,
JohnM
Ok so I figured it out, It was me being dumb!
Proper format is indeed:
Variable: var1=23456|var2=246810
which I would have sworn I tried and it failed but, I started at
the beginning again and voila!
JohnM
Un top posting for readability
On 8/16/2011 8:33 AM, Amol Vedak wrote:
Hi John,
I kind of facing the same problem that you were facing.
I am using similar configuration as you are for asterisk.
I am using java-asterisk library to communicate with asterisk.
In my code I am setting two variables (PIN, MREQID) and trying to
access them in dialplan (dialplan shown below).
When I send command to Asterisk to orginate, I get following result
(result shown below). I am wondering how should get access to
individual variable data. I was wondering if I should use Set(var,x,y)
method to pull out the part which is necessary for me. But wasnt sure
if thats the right way.
RESULT
-- Executing [login@authcheckrohan:5]
Set(SIP/softphonerohan-0060, PIN=3408|MREQID=1) in new stack
[Aug 16 17:53:06] WARNING[15739]: pbx.c:1344 pbx_exec: The application
delimiter is now the comma, not the pipe. Did you forget to convert
your dialplan? (Set(PIN=3408|MREQID=1))
-- Executing [login@authcheckrohan:6]
Set(SIP/softphonerohan-0060, MREQID=) in new stack
DIALPLAN
exten = login,1,NoOp();
;exten = login,n,SayNumber(${PIN})
exten = login,n,Set(E=${PIN})
exten = login,n,Verbose(${${E}_PIN})
exten = login,n,Verbose(${E})
exten = login,n,Set(PIN=${PIN})
exten = login,n,Set(MREQID=${MREQID})
exten = login,n,SayNumber(${MREQID})
Have you done it differently?
Thanks Regards,
Amol
I am connecting to the AMI from a C# app that was built by others but I
am using the same information and format as is used for a standard
telnet connection. What eneded up working is sending Variable:
var1=|var2=x|var3= as the last element(I do not think it is
important that it be last though). This is how it ended up in C# after
having established the connection:
//Tell asterisk who to call and to connect them to the IVR
clientSocket.Send(Encoding.ASCII.GetBytes(Action:
Originate\r\nChannel: sip/ + phoneNum1 + @provider\r\nMaxRetries:
2\r\nRetryTime: 60\r\nWaitTime: 30\r\nContext: the contextl\r\nExten:
the exten\r\nPriority: 1\r\nCallerid: XX\r\nAccount: CDR
Accountcode\r\nVariable: var1= + memberNum +|var2= + phoneNum1 +
|var3= + phoneNum2 + \r\n\r\n));
Then in my Dialplan I just use ${var1} ,${var2} , and ${var3} where I
need them.
Hope this helps.
JohnM
--
_
-- 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] AMI Commands - not working as Expected, Maybe???
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI. I have tried all
sorts of combinations,entering them after making a connection to ami
through telnet, of the many available examples on voip-info.org such as:
Action: Originate
Channel: sip/xx@xxx
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: test1
Exten: acs1
Priority: 1
CallerID: xx
Account: MyTest
Command: Set(var1=123456)
Command: Set(var2=54321)
also tried:
Var:
Variable:
SetVar:
Each individually for the two variables I need and both on the same line
separated by a | or a ,
Always when I hit return twice to give the \r\n\r\n The call is
successful but where I have
exten = acs1,n,NoOp(Vars = ${var1}, ${var2});
in my dialplan what I get is:
[2011-08-15 17:20:28] -- Executing [acs1@test1:2]
NoOp(SIP/xxx-0451, Vars = | ) in new stack
Obviously not what I was hoping for.
Any help would be greatly appreciated.
TIA,
JohnM
--
_
-- 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] AMI Commands - not working as Expected, Maybe???
On 8/15/2011 5:48 PM, john Millican wrote:
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI. I have tried all
sorts of combinations,entering them after making a connection to ami
through telnet, of the many available examples on voip-info.org such as:
Action: Originate
Channel: sip/xx@xxx
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: test1
Exten: acs1
Priority: 1
CallerID: xx
Account: MyTest
Command: Set(var1=123456)
Command: Set(var2=54321)
also tried:
Var:
Variable:
SetVar:
Each individually for the two variables I need and both on the same
line separated by a | or a ,
Always when I hit return twice to give the \r\n\r\n The call is
successful but where I have
exten = acs1,n,NoOp(Vars = ${var1}, ${var2});
in my dialplan what I get is:
[2011-08-15 17:20:28] -- Executing [acs1@test1:2]
NoOp(SIP/xxx-0451, Vars = | ) in new stack
Obviously not what I was hoping for.
Any help would be greatly appreciated.
TIA,
JohnM
Ok so I figured it out, It was me being dumb!
Proper format is indeed:
Variable: var1=23456|var2=246810
which I would have sworn I tried and it failed but, I started at the
beginning again and voila!
JohnM
--
_
-- 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] Securing Asterisk
On 7/28/2011 11:31 AM, Bruce B wrote: Hmmm, if alwaysauthreject is already breaking RFC rules then why not break another rule for the greater good? It would only add another layer of security. Maybe: *alwaysregreject=yes* * * *To drop SIP packets for both unauthorized registers and anonymous calls. Keep it off by default and then allow users to turn it on if they want to. To be fair to OP, using Asterisk with open ports to the world is a legit use of Asterisk even if most of us don't employ it that way or use it solely with closed networks (VPN, etc...). There are many people who would benefit from a security feature that would simply ignore unauthorized registers and anonymous calls. OP is suggesting an improvement to Asterisk; maybe people should weigh options and see if it's time to act more on the security side or not. There is no question that if a hacker knows there is a SIP server then they will keep the IP on the list for later use or share it with colleagues even if it seems secure right now. A DDoS is always a possibility and that you can't save yourself from at all. Right now the situation is more like this: *Knock Knock:* *Owner: *Whose there? *Thief:* This is Mr. X from China, and I am here to steal your TV. *Owner: *Hi, I am James Smith, 45, 190lbs and I have a nice laptop as well but I am home now and I can't let you in. *Thief (laughing):* No problem, I will come back at midnight when you are sleeping :-) - Bruce What I didn't tell you Mr thief is I sleep very lightly, Have a shotgun, a shovel and 20 acres of back yard and I know how to use all three! Why is there such an aversion to using the right tool for the job? Asterisk is not the security tool it is the voice tool! JohnM -- _ -- 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] Looking for actual user opinions on Telephony card
Hello all, Just hoping to get some opinions from folks that have actually used the Rhino R4FXO-EC. Looking for user experiences, good or bad. This looks like a nice unit and I have a need for exactly this config, 4FXO and EC TIA, JohnM -- _ -- 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] deadagi on v1.4.xx
Hello all,
I have a perl script that updates a M$ SQL DB based on an ivr that is
run on asterisk.
When it runs as a normal agi, it works great.
when run as a DeadAGI it does not work.
When i execute the script from h channel withDeadAGI and agi debug on i get:
[2010-12-20 01:08:54] -- Launched AGI Script
/var/lib/asterisk/agi-bin/insert_10day_var.pl
[2010-12-20 01:08:54] AGI Tx agi_request: insert_10day_var.pl
[2010-12-20 01:08:54] AGI Tx agi_channel: SIP/
[2010-12-20 01:08:54] AGI Tx agi_language: en
[2010-12-20 01:08:54] AGI Tx agi_type: SIP
[2010-12-20 01:08:54] AGI Tx agi_uniqueid: 1292825277.243
[2010-12-20 01:08:54] AGI Tx agi_callerid: appropriate caller id
[2010-12-20 01:08:54] AGI Tx agi_calleridname: appropriate name
[2010-12-20 01:08:54] AGI Tx agi_callingpres: 0
[2010-12-20 01:08:54] AGI Tx agi_callingani2: 0
[2010-12-20 01:08:54] AGI Tx agi_callington: 0
[2010-12-20 01:08:54] AGI Tx agi_callingtns: 0
[2010-12-20 01:08:54] AGI Tx agi_dnid: unknown
[2010-12-20 01:08:54] AGI Tx agi_rdnis: unknown
[2010-12-20 01:08:54] AGI Tx agi_context: my ivr context
[2010-12-20 01:08:54] AGI Tx agi_extension: h
[2010-12-20 01:08:54] AGI Tx agi_priority: 3
[2010-12-20 01:08:54] AGI Tx agi_enhanced: 0.0
[2010-12-20 01:08:54] AGI Tx agi_accountcode: the account code set
in call file
[2010-12-20 01:08:54] AGI Tx
[2010-12-20 01:08:54] -- AGI Script insert_10day_var.pl completed,
returning 0
Which is identical to the debug output when it work from the live
channel AGI
but I do not get the data in the db as it does when run by hand.
I am very tired, frustrated and have been googleing my butt of, no luck.
I know the script is getting the vars in as I had mistakenly left a
print statement in, which of course caused the script to bail but it
showed the correct info before it failed.
Could it be that even though it is running DeadAGI that there is a
sighup killing the script? if any suggestions on what to do about it?
AT: http://www.voip-info.org/wiki/view/Asterisk+cmd+DeadAGI
I did see that in 1.2 even on a deadagi I might have to catch the sighup
and it said
Your script will have to block SIGHUP signals, which you can do like so:
Perl:
$SIG{HUP} = IGNORE
I tried this and now at least I get a status after the DeadAGI returns,
which it did not get with out it. Although the status is FAILURE.
Any suggestions?
Thanks,
JohnM
--
_
-- 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] Maybe a little OT??--- Obtaining DIDs in Hyderabad, India
Hello, I originally thought I should post to the biz list but I am not looking for offers of DID's, I am looking for actual user experiences/information on obtaining a DID for an Office I am working with in Hyderabad, India. Can anyone offer recommendations based on personal experience of where I might be able to obtain said DID? This will be 90% inbound traffic and only within India. If anyone feels strongly that I should have indeed posted this to the biz list, please accept my apologies but, I felt I would get more pertinent info here. Based on this info I can then go to the biz list and ask for offers or straight to the discussed provider. Thank You, JohnM -- _ -- 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] Disa not fully bridging outbound call
John Millican wrote: Hello, I have a situation where a remote worker dials in to the asterisk server, enters the secret code, then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] -- Moving call from channel 21 to channel 2 [Jan 25 17:51:40] -- Zap/0:2-1 answered Zap/1-1 [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] -- Native bridging Zap/1-1 and Zap/0:2-1 [Jan 25 17:51:49] -- Channel 0/1, span 1 got hangup request, cause 16 [Jan 25 17:51:49] -- Hungup 'Zap/0:2-1' [Jan 25 17:51:49] == Spawn extension (from-inside-redir, 16037649936, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Executing [...@from-inside-redir:1] Hangup(Zap/1-1, ) in new stack [Jan 25 17:51:49] == Spawn extension (from-inside-redir, h, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Hungup 'Zap/1-1' [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:8615 pri_fixup_principle: Call specified, but not found? [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:9759 pri_dchannel: Hangup on bad channel 0/2 on span 1 This says it is using DAHDI but it is actually still Zaptel as I have not had much success getting DAHDI to work on OpenSuSE, but that is another post for a later date. Any help is greatly appreciated. Thank You As an FYI reply to my own post I was able to clear up the issue by rmmod and restart of zaptel. Not what I would call a good solution but it worked. Does not tell me what caused the problem but at least the customer is happy for now. JohnM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disa not fully bridging outbound call
Hello, I have a situation where a remote worker dials in to the asterisk server, enters the secret code, then dials out via Disa on a PRI. This was all working great until this morning. Now the calls is placed out, connected but there is no voice from/to either phone. This is what shows on the CLI when the calls is bridged at a verbose of 4 and a debug of 1: [Jan 25 17:51:40] -- Moving call from channel 21 to channel 2 [Jan 25 17:51:40] -- Zap/0:2-1 answered Zap/1-1 [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] WARNING[5563]: chan_dahdi.c:1594 conf_add: Failed to add 20 to conference 9/1: Invalid argument [Jan 25 17:51:40] -- Native bridging Zap/1-1 and Zap/0:2-1 [Jan 25 17:51:49] -- Channel 0/1, span 1 got hangup request, cause 16 [Jan 25 17:51:49] -- Hungup 'Zap/0:2-1' [Jan 25 17:51:49] == Spawn extension (from-inside-redir, 16037649936, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Executing [...@from-inside-redir:1] Hangup(Zap/1-1, ) in new stack [Jan 25 17:51:49] == Spawn extension (from-inside-redir, h, 1) exited non-zero on 'Zap/1-1' [Jan 25 17:51:49] -- Hungup 'Zap/1-1' [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:8615 pri_fixup_principle: Call specified, but not found? [Jan 25 17:51:49] WARNING[5412]: chan_dahdi.c:9759 pri_dchannel: Hangup on bad channel 0/2 on span 1 This says it is using DAHDI but it is actually still Zaptel as I have not had much success getting DAHDI to work on OpenSuSE, but that is another post for a later date. Any help is greatly appreciated. Thank You -- JohnM -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send the same message to list of users
David Gibbons wrote: snip Customers in Europe all have mobile phones, while senders in North America rarely have them ( they have answering machines, though ). /snip What planet/year are you/your clients living on/in? I don’t know anyone who doesn’t have a mobile. Maybe it’s just that they call it a cell phone instead of a mobile J How could anyone possible consider themselves a serious business person without a cell phone? That’s laughable. -Dave Planet Earth, third form the sun, Gregorian calender year of 2009 There are still vary large parts of the US that do not have cell coverage. So whether you want/need a cell/mobile or not is irrelevant. It is not so laughable when you don't have a choice. JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Joe Greco wrote: Sorry, I can't resist. How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. How about leaving member chastisement to the sponsor of the list? That's unlikely to happen in most cases. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Yes. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Be charitable and kind. Have a nice day. There's absolutely something to be said for that. On the other hand, there is also something to be said for making people exhaust the available resources prior to solving their problems for them. You can even be charitable and kind while doing so... Back in the '90's, I knew a really bright guy who knew Windows and Novell inside and out. He was just learning UNIXy stuff (FreeBSD in particular) and he was discovering that there was a lot of application for the stuff. He would frequently approach me, desperately seeking an answer to some general problem of some sort. I would typically give relatively vague answers, ending up essentially with a figure it out yourself. This frustrated him to no end, but he would do so. Later, he would come to me, almost always with a workable solution, at which point we would often discuss the ins and outs of several different options. His solution wasn't always the *best*, but it would always serve as a foundation for the rest. Years later, he thanked me. At the time, he didn't really appreciate what I was doing and didn't see the bigger picture. Looking back on it, I think he saw that I had always tried to aim him in a sensible direction before shoving him off on his own to figure it out. He eventually grew confident enough and capable enough that he would no longer need to ask for help. I can fix your problems for you, or I can teach you to be self- sufficient... which one is doing you more of a favor? It may seem more charitable and kind to simply give someone answers, but I do not think it actually is, at least in this sort of situation. As for the original poster? It's my impression, reading in between the lines, that he probably hasn't tried that hard. Asterisk on Linux is pretty straightforward, and MOH is probably not that rough to get running. On FreeBSD? That's a different thing. Bleh. But it's still better to do it on-list rather than selecting someone at random to go and bother. I don't think anyone will prevent you from being charitable and kind by providing answers to the guy's questions on the list though. ... JG Slightly paraphrasing a very old and wise saying: Give a man a fish, he eats for a day. Teach him how to fish, he eats for a lifetime. -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A little OT but need an opinion on Aastra 57i CT
Hello All, I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover up to 300,000 square feet. I am finding this hard to accept. I was also wondering about the secure WDCT cordless technology Could this be a form of DECT? Any one using these that can shed some lite? Thanks. JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] chanspy and DISA
Steve Edwards wrote:
Steve Edwards wrote:
Is the manager or are the agents using disa()?
How about:
exten = *,n,set(SPYGROUP=ALLOW-SPYING)
for the agents and:
exten = *,n,chanspy(,g(ALLOW-SPYING))
the manager?
On Tue, 29 Sep 2009, John Millican wrote:
The manager wants to be able to spy on agents who dial through the PBX
from their homes. Currently the agents dial the main number, use the
secret code to get to authenticate and DISA, and then dial back out
for their sales calls. I have chanspy working great on all internal
phones/extensions use group to limit who can spy and who can not. It not
so much to allow spying it is finding the correct channel to spy on for
the remote users.
How about something like these snippets:
[i](!)
exten = i,1,goto(${CONTEXT},s,1)
[s](!)
exten = s,1,verbose(1,[${CONTEXT}:${EXTEN}])
[home-agent-login](i,s)
exten = s,n,read(AGENT-ID,enter-agent-number)
exten = s,n,set(SPYGROUP=${AGENT-ID})
.
.
.
[supervisor-login](i,s)
exten = s,n,read(AGENT-ID,enter-agent-number)
exten = s,n,chanspy(,g(${AGENT-ID}))
exten = s,n,goto(s,1)
.
.
.
Thank you very much for this.
With a little tweaking it worked great, since each remote workers
callerid is matched before going to authenticate I just set the spy
group so the remote guys don't have a choice and now the manager has a
known group of one for each remote worker.
Thanks again for the help
JohnM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
[asterisk-users] chanspy and DISA
Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a way to find the channel to barge on that would be usable by NON tech people? Any thoughts? TIA, JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] chanspy and DISA
Steve Edwards wrote:
On Tue, 29 Sep 2009, John Millican wrote:
I have a request for remote users to be able to dial through the system
so that the sales managers can barge/chanspy on the sales force. I have
the DISA part working with authentication(rather straight forward) but
what I can not figure out is how to enable the supervisors to be able to
barge on these calls. Is there a way to find the channel to barge on
that would be usable by NON tech people?
How do you see this working? I'm guessing the manager would like to either
key in an agent ID number or be able to step through agents?
The chanspy() g option may be part of your solution.
g(grp) - Match only channels where their ${SPYGROUP} variable is set
to 'grp'.
Exactly, the problem is I can not determine the channel that DISA
receives or places the call on. Is there a way to set this in the dial
plan? Or am I just missing something simple? It was suggested to use
the AMI and present the info as a web page but this will require
retraining the manager, as we all know this is a notoriously difficult
process.
JohnM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] chanspy and DISA
Steve Edwards wrote:
On Tue, 29 Sep 2009, John Millican wrote:
I have a request for remote users to be able to dial through the system
so that the sales managers can barge/chanspy on the sales force. I have
the DISA part working with authentication(rather straight forward) but
what I can not figure out is how to enable the supervisors to be able to
barge on these calls. Is there a way to find the channel to barge on
that would be usable by NON tech people?
Steve Edwards wrote:
How do you see this working? I'm guessing the manager would like to either
key in an agent ID number or be able to step through agents?
The chanspy() g option may be part of your solution.
g(grp) - Match only channels where their ${SPYGROUP} variable is set
to 'grp'.
On Tue, 29 Sep 2009, John Millican wrote:
Exactly, the problem is I can not determine the channel that DISA
receives or places the call on. Is there a way to set this in the dial
plan? Or am I just missing something simple? It was suggested to use
the AMI and present the info as a web page but this will require
retraining the manager, as we all know this is a notoriously difficult
process.
Is the manager or are the agents using disa()?
How about:
exten = *,n,set(SPYGROUP=ALLOW-SPYING)
for the agents and:
exten = *,n,chanspy(,g(ALLOW-SPYING))
the manager?
The manager wants to be able to spy on agents who dial through the PBX
from their homes. Currently the agents dial the main number, use the
secret code to get to authenticate and DISA, and then dial back out
for their sales calls.
I have chanspy working great on all internal phones/extensions use group
to limit who can spy and who can not. It not so much to allow spying it
is finding the correct channel to spy on for the remote users.
JohnM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] [SPAM] RE: dCAP Exam
C. Savinovich wrote: What about if I use the browser from my cellular phone? CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno Sent: Wednesday, September 16, 2009 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dCAP Exam I believe the administrator can see what is on your screen with screen with those screen sharing stuff, this makes it harder a lil bit, and www.boratproxy.com becomes useless in that case. On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Wed, Sep 16, 2009 at 11:26 AM, Tilghman Lesher tles...@digium.com wrote: On Tuesday 15 September 2009 20:14:32 Neeraj Chand wrote: Hmm...so by open book, that means access to the internet? Possible to get own notes ? Yes, you have access to the Internet, but your access is proxied, and the administrator of the test can see everything that you access. So it's best for you stick with only general guides and not look for crib notes. If your test proctor believes you cheated, you fail. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org Just tunnel your HTTP traffic over an SSH link and go to some dCAP brain dump sites. Or go to www.boratproxy.com and confuse their proxy. ah too fun. I can't resist: After having taken many MS and Cisco tests in the past, it would seem rather apparent to me that for the dcap, as with any other test, if you know what you are doing you are all set and you don't have to try and find ways to cheat that don't look like cheating. Disclaimer: I have not taken the dcap test yet! JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Aastra 51i and PAP2T behind NAT
OK this is the RTFM question of the day but I need a sanity check. I have 2 Astra 51i phone and a linksys PAP2 on a single DSL connection. 2 Aastra 51i-| |-NAT on dsl moden--(Internet)--Asterisk PAP2t| The DSL modem/router which has QOS set for the src and dest to the * box the PAP2 has both lines registered @ ports 5060 and 5061 and work like a charm. one of the the aastra's registered at port 1025 worked all day but the showed no service and lost registration over night sometime. this happens with much regularity. I am looking for the docs on these phones to see if they have a NAT keep alive option. Does this sound like a reasonable place to start for a solution? Thanks in advance -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] probably an rtfm but... need to dial out to 2 PSTN lines from AMI
Hello all, I have a need to be able to use the originate AMI command to dial out to the PSTN, have that person answer and then have the second PSTN connection dialed out. I have tried to use: Action: Originate Channel: sip/number@provider Context: default Exten: othernumber Priority: 1 Timeout: 3 This does not dial the number through the provider, actually, it seems that the number never gets passed to the provider. I suppose I could create a dummy sip exten but it would have to be one that had no device attached and I am unclear on how to do that. Any Sugestion on either method? TIA -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] probably an rtfm but... need to dial out to 2 PSTNlines from AMI
Danny Nicholas wrote: users.conf [108] username = 108 transfer = yes mailbox = 108 call-limit = 100 fullname = General Messages registersip = no host = dynamic callgroup = 1 context = DLPN_DialPlan1 cid_number = 108 hasvoicemail = yes vmsecret = 1234 email = du...@dummy.com threewaycalling = no hasdirectory = no callwaiting = no hasmanager = no hasagent = no hassip = yes hasiax = no secret = nat = yes canreinvite = no dtmfmode = rfc2833 insecure = no pickupgroup = 1 disallow = all allow = ulaw,gsm autoprov = no label = macaddress = linenumber = 1 no entry in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Millican Sent: Thursday, May 28, 2009 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] probably an rtfm but... need to dial out to 2 PSTNlines from AMI Hello all, I have a need to be able to use the originate AMI command to dial out to the PSTN, have that person answer and then have the second PSTN connection dialed out. I have tried to use: Action: Originate Channel: sip/number@provider Context: default Exten: othernumber Priority: 1 Timeout: 3 This does not dial the number through the provider, actually, it seems that the number never gets passed to the provider. I suppose I could create a dummy sip exten but it would have to be one that had no device attached and I am unclear on how to do that. Any Sugestion on either method? TIA Thanks for the info Danny. I also found while doing more reading that I can use Action: Originate Channel: local/1...@mynewcontext Context: default Exten: othernumber Priority: 1 Timeout: 3 and then setup a context in the dial plan that dial out to the needed number. I new as soon as I sent the question something rtfm ish would hit me Thanks again -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Dialplan Digitmaps
Justin Phelps wrote: I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from the old configs though, and I need some help figuring out why these expressions lock up the new phone. Old Configs digitmap dialplan.digitmap=[2-9]11|[*]xx|891xxx|[1-7]xxx|8[0-46-8]xx|8500|851xxx|9,1[2-9]x|9,xxxT ^^^ dialplan.digitmap.timeOut=3/ Template from Polycom digitmap dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2-9]xxxT dialplan.digitmap.timeOut=3|3|3|3|3|3/ Anyone have any insight or suggestions on this issue, and on upgrading Polycom configs in general? Do those certain numbers start with a 1 through 7? You may have to put a T in the above marked section and add to the series of dialplan.digitmap.timeOut. Just to keep myself from getting confused i usually put in the full string of time outs, needed or not. -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Dialplan Digitmaps
Justin Phelps wrote: digitmap dialplan.digit map=[2-9]11T|[*]xxT|891xxxT|[1-7]xxxT|8[0-46-8]xxT|8500T|851xxxT|9,1[2-9]xT|9,xxxT dialplan.digitmap.timeOut=3|3|3|3|3|3|3|3|3/ Do the above changes look in line with common practice JohnM? Short Answer: They do. Longer answer, You only need to put the T in the dialplan.digit map where you might need to wait to be sure the user is finished dialing, it will not hurt to have it there. This way if you have extensions like 1234 and 12345 you can use something like |123X.T| bad example but you get the idea. This will wait for the amount of seconds that you have in the dialplan.digitmap.timeOut for that section after the user has dialed 4 digits to be sure that they are not going to dial the 5th digit. I put the time in for all section in the dialplan.digitmap.timeOut sect whether iput the T in the dialplan.digit map section or not just to make it easer for my pea brain to follow what time out corralates to which dial section. You may not want to wait for 3 seconds in all dial condition, you may put a 1 or a 2 second wait for some and a 3 or for second wait for others. I hope this has not confused matters any more. -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall - canunicall affect zaptel commands?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Tuesday, April 07, 2009 6:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall - canunicall affect zaptel commands? The message includes a host of irrelevant and relevant information. The question is not clear. It is a horrible piece of top-posting mess. Please provide the relevant configuration again and clarify your answer. What hardware do you have? What connections do you have? Are they working OK? Generally chan_zap and chan_unicall should not handle the same spans -- 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 Forwarded message -- From: Juan Carlos Huerta juancarloshue...@gmail.com Date: 07-abr-2009 13:41 Subject: Re: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands? To: asterisk...@lists.digium.com Please wirte to asterisk-users@lists.digium.com to get help about this problem. Juan Carlos ~ Lo que no te mata te fortalece ~ On Tue, Apr 7, 2009 at 1:35 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: Thanks, Juan Carlos. Yes, I still can t type the commands in the asterisk console . I dont have dial tone for FXS ports . Regards. Giovanni. 2009/4/7, Juan Carlos Huerta juancarloshue...@gmail.com: And what is the result? still with zap commands problem? Juan Carlos ~ Lo que no te mata te fortalece ~ On Tue, Apr 7, 2009 at 12:49 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: I did it: elastix*CLI load chan_zap.so The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata_additional.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found -- Registered channel 63, FXO Kewlstart signalling -- Registered channel 64, FXO Kewlstart signalling -- Registered channel 65, FXO Kewlstart signalling -- Registered channel 66, FXO Kewlstart signalling -- Registered channel 67, FXO Kewlstart signalling -- Registered channel 68, FXO Kewlstart signalling -- Registered channel 68, FXO Kewlstart signalling -- Registered channel 69, FXO Kewlstart signalling -- Registered channel 70, FXO Kewlstart signalling -- Registered channel 71, FXO Kewlstart signalling -- Registered channel 72, FXO Kewlstart signalling -- Registered channel 73, FXO Kewlstart signalling -- Registered channel 74, FXO Kewlstart signalling -- Registered channel 75, FXO Kewlstart signalling -- Registered channel 76, FXO Kewlstart signalling -- Registered channel 77, FXO Kewlstart signalling -- Registered channel 78, FXO Kewlstart signalling elastix*CLI Thank you Juan Giovanni 2009/4/7, Juan Carlos Huerta juancarloshue...@gmail.com: Try to load the chan_zap.so module manually to see if you get some error. Juan Carlos ~ Lo que no te mata te fortalece ~ On Tue, Apr 7, 2009 at 12:44 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: I'm using Elastix 1.1-8 with: asterisk-1.4.19 spandsp-0.0.4 unicall-0.0.5pre1 zaptel-1.4.9.2 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 Regards, Giovanni 2009/4/7, Giovanny Magallanes gmagalla...@gmail.com: Ok, Thanks. The chan_zap commands are unavailable, but unicall commands and my E1 MFC/R2 are OK. Giovanni 2009/4/7, Moises Silva moises.si...@gmail.com: I don't understand your problem. And no, unicall has nothing to do with chan_zap.so commands. Please, in the future, don't hijack threads, open a new thread for your discussion. This time I changed the subject already. Moy On Tue, Apr 7, 2009 at 1:08 AM, Giovanny Magallanes gmagalla...@gmail.com wrote: Hi, Guys. I did not type any zaptel commands in asterisk console. I have installed Elastix 1.1-8 with unicall and it is working fine with a Digium TE212-P. [r...@elastix asterisk]# find -P / -name chan_zap.so /usr/lib/asterisk/modules/chan_zap.so /usr/src/astunicall-1.4.19-0.1/asterisk-1.4.19/channels/chan_zap.so [r...@elastix asterisk]# elastix*CLI load chan_zap.so == Parsing '/etc/asterisk/zapata.conf': Found == Parsing '/etc/asterisk/zapata_additional.conf': Found == Parsing '/etc/asterisk/zapata-channels.conf': Found -- Registered channel 63, FXO Kewlstart signalling -- Registered channel 64, FXO Kewlstart signalling -- Registered channel 65,
Re: [asterisk-users] Area code 757 Car warranty calls
Jon Pounder wrote: Cary Fitch wrote: The problem has two prongs - first we are in control of our own landlines and can use asterisk to screen whatever crap we wish before disturbing a real user or allowing a vm to get stored (but it would be nice not to have to). The other issue is we are not for the most part in any kind of control situation of our cellphones, and there is no way to stop that ring from happening and once it does it either needs to be answered or a vm dealt with. This is where the bigger players need to start living up to their responsibilities and not just ignore the problem. Well it will get me off my rant in this forum. Isn't that worth something? Seriously, as users some of us have one 2 line system and others are running multiple systems, absorbing hundreds of thousands of calls a day. Where the %#! warranty calls are coming from or not coming from is useful info. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Friday, March 20, 2009 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Area code 757 Car warranty calls This information appears to be relevant, but useless? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, March 20, 2009 10:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Area code 757 Car warranty calls On Friday 20 March 2009 00:38:45 Cary Fitch wrote: Sure if you can get up stream carriers to cooperate. Just follow the CDRs. But short of a subpoena... or enlightened self interest, like the calls take down a tandem.. (not likely). We could loop the calls back to get ATT's attention, but they would just complain about the loop, not trace them back to the source. Nothing official, but if these are the same clowns who called me earlier this month (and who I filed a complaint on at the DNC registry), then changing their area code may have been a ploy to avoid more complaints. Here is some relevant information on that number: http://whocalled.us/lookup/7025200085 I realize that finding these (insert foul and derogatory expletive), but if we do maybe a public whipping with a cat-o-nine tails on the six O'Clock news? It is my phone, I pay for the service I should not have to answer (or even filter out) calls from some idiot that has absolutely no business calling me in the first place. There are many other avenues of advertising that are not invasive of my privacy and do not require me to pay for the call in the case of a cell phone number. Maybe sending a bill to some of these jerks for all the cell calls they have made will hit them where it hurts. Just my opinion JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No hardware timing source found in /proc/dahdi
Shaun Ruffell wrote:
John Millican wrote:
Well,
lsmod | grep hisax returns nothing
plain lsmod:
Module Size Used by
dahdi_dummy22472 0
dahdi 215776 1 dahdi_dummy
crc_ccitt 18944 1 dahdi
af_packet 57100 2
snd_pcm_oss67456 0
snd_mixer_oss 34176 1 snd_pcm_oss
snd_seq74992 0
snd_seq_device 25620 1 snd_seq
vmnet 72992 3
parport_pc 58456 0
parport56588 1 parport_pc
vmmon 158908 0
sunrpc198600 1
iptable_filter 19840 0
ip_tables 37848 1 iptable_filter
ip6table_filter19584 0
ip6_tables 31944 1 ip6table_filter
x_tables 37000 2 ip_tables,ip6_tables
ipv6 372344 29
cpufreq_conservative24968 0
cpufreq_userspace 23680 0
cpufreq_powersave 18560 0
powernow_k831504 0
apparmor 58672 0
loop 36356 0
dm_mod 77152 0
ohci1394 51272 0
ieee1394 115800 1 ohci1394
i2c_nforce222784 0
snd_hda_intel 368804 0
i2c_core 43648 1 i2c_nforce2
snd_pcm 108680 2 snd_pcm_oss,snd_hda_intel
snd_timer 42632 2 snd_seq,snd_pcm
snd84984 7
snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm,snd_timer
k8temp 22656 0
hwmon 20232 1 k8temp
button 26400 0
usblp 30976 0
forcedeth 65416 0
rtc_cmos 25016 0
rtc_core 38156 1 rtc_cmos
rtc_lib19968 1 rtc_core
sr_mod 33444 0
cdrom 52392 1 sr_mod
usb_storage 102816 0
soundcore 25360 1 snd
snd_page_alloc 27280 2 snd_hda_intel,snd_pcm
ide_core 165648 1 usb_storage
sg 53304 0
usbhid 58160 0
hid43776 1 usbhid
ff_memless 22536 1 usbhid
sd_mod 45824 6
ohci_hcd 38020 0
ehci_hcd 50572 0
usbcore 155560 6 usblp,usb_storage,usbhid,ohci_hcd,ehci_hcd
edd26760 0
ext3 156688 3
mbcache26248 1 ext3
jbd89192 1 ext3
fan22792 0
sata_nv38404 4
pata_amd 31876 0
libata164096 2 sata_nv,pata_amd
scsi_mod 176536 5 sr_mod,usb_storage,sg,sd_mod,libata
thermal34576 0
processor 59592 2 powernow_k8,thermal
Looking at the lsmod output, it appears that the wctdm module is not
loaded. So either the /etc/dahdi/modules has the wctdm module commented
out, or something is wrong with the /etc/init.d/dahdi that it isn't
viewing that file.
If you unload all the drivers ('/etc/init.d/dahdi stop') and make sure
they are unloaded ('lsmod | grep dahdi' should not show any output) then
just load the wctdm driver ('modprobe wctdm'), and then what does dmesg
show?
Well that did it. I guess I will have to modify /etc/init.d/dahdi to
only modprobe wctdm for now and run with it. wctdm was the only module
that was not commented out in /etc/dahdi/modules so it must be as you
said the /etc/init.d/dahdi was not reading the file as ity should. I
will look into what is happening there.
dmessg output:
dahdi: Telephony Interface Unloaded
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.1.0.4
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
dahdi_echocan_mg2: Registered echo canceler 'MG2'
dahdi: Registered tone zone 0 (United States / North America)
dahdi_cfg output:
DAHDI Tools Version - 2.1.0.2
DAHDI Version: 2.1.0.4
Echo Canceller(s): MG2
Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
4 channels to configure.
Changing signalling on channel 1 from Unused to FXO Kewlstart
Setting echocan for channel 1 to mg2
Changing signalling on channel 2 from Unused to FXO Kewlstart
Setting echocan for channel 2 to mg2
Changing signalling on channel 3 from Unused to FXS Kewlstart
Setting echocan for channel 3 to mg2
Changing signalling on channel 4 from Unused to FXS Kewlstart
Setting echocan for channel 4 to mg2
Thank you very much for your
[asterisk-users] No hardware timing source found in /proc/dahdi
Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAAGGH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get: install -D dahdi.init /etc/init.d/dahdi /sbin/chkconfig --add dahdi dahdi 0:off 1:off 2:on 3:on 4:on 5:on 6:off DAHDI has been configured. If you have any DAHDI hardware it is now recommended you edit /etc/dahdi/modules in order to load support for only the DAHDI hardware installed in this system. By default support for all DAHDI hardware is loaded at DAHDI start. I think that the DAHDI hardware you have on your system is: pci::03:08.0 wctdm- e159:0001 Wildcard TDM400P REV E/F so it is seeing the card /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=3,4 echocanceller=mg2,1-4 channels=1-4 /etc/dahdi/init.conf: MODULES=$MODULES wctdm /etc/dahdi/modules # Digium TDM400P: up to 4 analog ports wctdm # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: /usr/sbin/dahdi_cfg # /usr/sbin/dahdi_cfg - DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 4 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) *CLI dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO DAHDI_DUMMY/1 (source: RTC) 1UNCONFI 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) I am really hoping that I am just missing something stupid. Anyone have any suggestions? TIA JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No hardware timing source found in /proc/dahdi
Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: /usr/sbin/dahdi_cfg If the dmesg shows that the driver found the card and there were not any conflicts, and dahdi_dummy is still loaded, this could be the result of an open reference to the old /proc/dahdi directory. i.e., you can force this to happen if you 'modprobe dahdi cd /proc/dahdi modprobe -r dahdi /etc/init.d/dahdi start' Cheers, Shaun All I see in dmesg is: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 dahdi_dummy: RTC rate is 1024 -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No hardware timing source found in /proc/dahdi
Shaun Ruffell wrote: John Millican wrote: Shaun Ruffell wrote: John Millican wrote: # /etc/init.d/dahdi start Loading DAHDI hardware modules: wctdm: modprobe wctdm What is the output of the 'dmesg' command at this point? All I see in dmesg is: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 dahdi_dummy: RTC rate is 1024 Odds are there is another driver in your system then that is attaching to the tdm400 before the wctdm driver. I've seen where the hisax driver attaches first. Does 'lsmod | grep hisax' show that hisax is loaded? You can see what drivers may be configured for a board by looking at the /lib/modules/`uname -r`/modules.pcimap file. The tdm400 uses a vendorid of 0xe159 and a device id 0x0001. Search for any driver in the pcimap file that indicates support for that driver, and add it to the /etc/modprobe.d/blacklist file. Something like: blacklist hisax blacklist hisax_fcpcipnp Cheers, Shaun Well, lsmod | grep hisax returns nothing plain lsmod: Module Size Used by dahdi_dummy22472 0 dahdi 215776 1 dahdi_dummy crc_ccitt 18944 1 dahdi af_packet 57100 2 snd_pcm_oss67456 0 snd_mixer_oss 34176 1 snd_pcm_oss snd_seq74992 0 snd_seq_device 25620 1 snd_seq vmnet 72992 3 parport_pc 58456 0 parport56588 1 parport_pc vmmon 158908 0 sunrpc198600 1 iptable_filter 19840 0 ip_tables 37848 1 iptable_filter ip6table_filter19584 0 ip6_tables 31944 1 ip6table_filter x_tables 37000 2 ip_tables,ip6_tables ipv6 372344 29 cpufreq_conservative24968 0 cpufreq_userspace 23680 0 cpufreq_powersave 18560 0 powernow_k831504 0 apparmor 58672 0 loop 36356 0 dm_mod 77152 0 ohci1394 51272 0 ieee1394 115800 1 ohci1394 i2c_nforce222784 0 snd_hda_intel 368804 0 i2c_core 43648 1 i2c_nforce2 snd_pcm 108680 2 snd_pcm_oss,snd_hda_intel snd_timer 42632 2 snd_seq,snd_pcm snd84984 7 snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm,snd_timer k8temp 22656 0 hwmon 20232 1 k8temp button 26400 0 usblp 30976 0 forcedeth 65416 0 rtc_cmos 25016 0 rtc_core 38156 1 rtc_cmos rtc_lib19968 1 rtc_core sr_mod 33444 0 cdrom 52392 1 sr_mod usb_storage 102816 0 soundcore 25360 1 snd snd_page_alloc 27280 2 snd_hda_intel,snd_pcm ide_core 165648 1 usb_storage sg 53304 0 usbhid 58160 0 hid43776 1 usbhid ff_memless 22536 1 usbhid sd_mod 45824 6 ohci_hcd 38020 0 ehci_hcd 50572 0 usbcore 155560 6 usblp,usb_storage,usbhid,ohci_hcd,ehci_hcd edd26760 0 ext3 156688 3 mbcache26248 1 ext3 jbd89192 1 ext3 fan22792 0 sata_nv38404 4 pata_amd 31876 0 libata164096 2 sata_nv,pata_amd scsi_mod 176536 5 sr_mod,usb_storage,sg,sd_mod,libata thermal34576 0 processor 59592 2 powernow_k8,thermal in /etc/modprobe.d/blacklist there is already: # ISDN modules are load from /lib/udev/isdn.sh snip a lot of unrelated blacklist hisax blacklist hisax_fcpcipnp blacklist hisax_st5481 less /lib/modules/`uname -r`/modules.pcimap | grep 0xe159 hisax0xe159 0x0002 0x 0x 0x 0x 0x0 hisax0xe159 0x0001 0x 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa159 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xe159 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb100 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb1d9 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb118 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xb119 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa9fd 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa8fd 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa800 0x 0x 0x 0x0 wctdm0xe159 0x0001 0xa801
Re: [asterisk-users] Aastra phones
Mike wrote: Hi, I`ve been toying with an Aastra phone (9143i) wondering if it could be a good alternative to to the more expensive Polycom phones. One thing which I can't figure out, although it certainly looks simple, is to update the firmware though FTP (not TFTP). I have set the ftp provisioning server in the Aastra phone, and put the firmware file 9143i.st in the root folder where the login/password pair ends up. Everything is entered correctly, or so it seems (works fine with my Polycoms). When I reboot the phone from the Web UI, it doesn't seem to take in the new firmware. But it does seem to download the (empty) aastra.cfg file (proving that the provisoning server settings are correct). What am I missing? I believe that the older firmware for the Aastra phone will only update from TFTP. I am not sure what rev level this changed at though. JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX?
Chris Bagnall wrote: I would suggest using OpenSIPS with Asterisk and bypass IAX all together for this particular application. If the users in question are often in hotels abroad, something like this may not solve the problem - I've noticed quite a few hotels are now blocking SIP traffic (presumably so as to encourage people to use the hideously overpriced phones in their rooms to make calls from). Your best bet might well be some low-cost IAX handsets for those users who are unable/unwilling to use softphones. I think Atcom make some IAX handsets - quality isn't great compared to the usual suspects (Cisco, Polycom, Snom, etc.), but they do work. Assuming the users all have Wi-Fi on their laptops, an alternative might be a simple VPN setup on the laptops, bridged to their Wi-Fi card running in AP mode, then use something like the SIP client on a Wi-Fi capable mobile phone or a Wi-Fi SIP phone. An OpenSIPS solution will take care of your traveler's NAT issues (and could handle the registrations) while you used Asterisk for voicemail and whatever else. I've personally used this type of general setup in the past with a great deal of success for remote offices and soft-phones on laptops. Not directly on-topic for this list, but I'd not heard of OpenSIPS before, so I had a look at the website. It looks to be a fork of OpenSER. Does that mean OpenSER development has slowed/ceased, or has the OpenSER project itself morphed into OpenSIPS? Regards, Chris via a quick google:OpenSER is now OpenSIPS www.opensips.org OpenSER continues via OpenSIPS A new name, same project -- JohnM ___ -- 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
Re: [asterisk-users] GotoIftime
Ira wrote: At 01:36 PM 7/30/2008, you wrote: Nhadie wrote: Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) GotoIfTime(22:00-23:59|*|30-31|jul?test,s,1) GotoIfTime(00:00-02:00|*|30-31|jul?test,s,1) Doug Does that leave a 1 minute or 1 second hole? Should be neither. Since the time frame allowed is in minutes there is no time between 23:59 and 00:00 Since it has to be one or the other. Now, if I assume that the time is converted from time_t to 24 hour format using something such as localtime or gmtime the result of this should be using only tm_min and tm_hour which would also mean there is no hole. THESE ARE ALL ASSUMPTIONS, I have not checked the code. -- JohnM ___ -- 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
Re: [asterisk-users] Callerid Woes
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
--
JohnM
___
-- 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
Re: [asterisk-users] Callerid Woes
John Koenig wrote:
I tried all of the suggestions, and still the callerid remains intact.
I guess at this point I am starting to wonder what bit of logic is being
run when I dial *8111XX...
Is there a way I can trace how a call is being processed within
asterisk? Or even see what I am sending to my VoIP terminating node?
John
John Millican wrote:
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To set an empty callerId
typing:
sip set debug peer peer_name
at the CLI will give you a bunch of information as to what is going on
with that peer
--
JohnM
___
-- 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
[asterisk-users] CallerId show with IP address appended
Hello, Asterisk 1.4.21.1 Well it seems like my month for questions. I have a situation where the CallerID num shows as [EMAIL PROTECTED](the ip of the asterisk box) on calls to any of the internal phones. This prevents the ability to dial out from the missed call list. I have not been able to find out why this is happing. To further confuse the issue when i register and extension to the public IP from outside the firewall I get only 16035551212 as the clid. I have several NoOps in the dial plan and they all show the clid as 16035551212, which is also what is in the cdr, but when it gets to the Polycom it has the IP appended. The phones are all polycoms but have also tested with x-lite and it gets the ip appended also. Any pointers as to where to look? JohnM ___ -- 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
[asterisk-users] Beep on transfer
Hello All, I have a request that I have not been able to figure out as yet. I need to be able to play a beep when a call is transfered via attended transfer. This is exactly what is in the bug tracker at: http://bugs.digium.com/view.php?id=3819 Has any one found a way, elegant ot otherwise, to make something such as this work? Thanks in advance for any help. -- JohnM ___ -- 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
[asterisk-users] READ application
Hello, Asterisk version 1.4.21.1 Can anybody tell me what I am doing wrong or why the Read application does not accept the # key as input? My read statement: exten = s,n,Read(uchoice|thankyouforcalling|3||1|1); In the prompt thankyouforcalling it says press pound for a company directory along with some press this digit for blah blah. If the caller presses # the read applications exits and says that the user entered nothing. Really strange that the app hears the DTMF, since it stops the prompt, but does nothing with it. Is it because Read exits with a # terminated string so it sees ## and just ignores it? If this is the case then maybe Background is the answer. But I am unable to get Background to accept more than a single digit and I need to be able to grab up to 3 digits or the # key. My background statement: exten = s,n,Background(thankyouforcalling|m||macro-jm-closed) I have tried this wityh and with out the m option, same results. Both of these are run in a macro. Thank you for any help. JohnM ___ -- 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
Re: [asterisk-users] READ application
Tilghman Lesher wrote: On Wednesday 09 July 2008 09:08:50 John Millican wrote: Can anybody tell me what I am doing wrong or why the Read application does not accept the # key as input? My read statement: exten = s,n,Read(uchoice|thankyouforcalling|3||1|1); In the prompt thankyouforcalling it says press pound for a company directory along with some press this digit for blah blah. If the caller presses # the read applications exits and says that the user entered nothing. Really strange that the app hears the DTMF, since it stops the prompt, but does nothing with it. Is it because Read exits with a # terminated string so it sees ## and just ignores it? If this is the case then maybe Background is the answer. But I am unable to get Background to accept more than a single digit and I need to be able to grab up to 3 digits or the # key. My background statement: exten = s,n,Background(thankyouforcalling|m||macro-jm-closed) I have tried this wityh and with out the m option, same results. Both of these are run in a macro. Anything running in a Macro matches new extensions in the place where the Macro was called from. Background always matches new extensions, as opposed to Read, which collects DTMF for a variable. If Background is only matching single-digit extensions, then you only have single-digit extensions in the calling context. Thank you Jared and Tilghman I do have single digit and multiple digit extensions in the macro and from core show application background I found that by using the context option: exten = s,n,Background(thankyouforcalling|||macro-jm-closed) ^^^ when calling background it will return to the macro from which it was called, which it does as i can get to the defined single digit extens such as exten = 3,1,do something exten = 4,1,do something which are ONLY defined in the macro but not the extensions defined such as: exten = _2XX,1,do something for my three digit internal phone extensions So, next try. Since Read resets the value of the variable to an empty string after the timeout if the caller does nothing or if the caller presses #, is there a way to test if the caller pressed # in Read to cause it to terminate or if the application simply timed out. If I can test for this i would be able to tell if the user pressed # or not since any other key press will insert a value into the variable. Problem is that if they do nothing or press # I get the empty variable so I cant simply use LEN and a GoToIf. Thanks again JohnM ___ -- 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
[asterisk-users] Read Background
Hello All,
Asterisk 1.4.20.1
SuSE 10.3
I have been building a dial plan and have run into some questions that I
have not been able to answer on Voip-info or google. I am trying to use
either Read or Background to gather user input to an IVR in a Macro. I
need to be able to branch based on the user entering the # key or up to
3 digits. When I use Background I get the # key but am only able to
collect one digit. If I use Read I can the three digits but the # key
is not recognized. The system knows that a key was pressed as it exits
the prompt immediately but it then spits out to the CLI that the user
entered nothing and the variable uchoice is in fact empty.
What I have had to do is use a combination of both Read and Background,
rather ugly in my opinion.
Macro snipit:
exten = s,n,Read(uchoice|outmessg/greeting|3||1|3);
exten = s,n,GoToIf($[${LEN(${uchoice})}0]?${uchoice},1);
exten = s,n,Background(outmessg/directory_rotary|m||macro-jm-in);
;have tried the above with and without the m option
exten = s,n,WaitExten(3);
This works but puts an unfortunate pause between the two prompts in
order to give the caller time to decide what they want to do. I can't
just test for empty/NULL to see if the user hit the # key as I need to
go to operator if the user does nothing which also leaves uchoice set to
nothing. I have set uchoice to 0 previously in the dialplan but when it
goes through read it gets reset to either the user entry of nothing if
the user does not press a key or presses #.
I would just not use the # key but the prompts were previously recorded
and I need to match prior functionality.
If I could get Read to recognize the # key or Background to accept more
than one digit I would be a much happier camper. Am I missing something?
Am in process of upgrading a test box to 1.4.21.1 to see if there is any
change, but hopes are not high.
Any help is much appreciated.
Thanks,
JohnM
___
-- 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
[asterisk-users] Asterisk and TDD
Hello all, I was just asked a question from a client that I have in regards to TTY/TDD telecommunications device for the deaf. I have read on voipinfo at http://www.voip-info.org/wiki/view/tdd+mode that back in Dec 2006 this was in alpha stage in Asterisk. There does not (in my limited searching) seam to be any other documentation. Is this in 1.4/1.6? Is anyone using it? How well does it work? TIA -- JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interrupting MOH
Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like please wait while we transfer your call. How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can be accomplished? The only way to do that currently is to implement the prompt within the MOH stream itself. Just off the top-o-my head(YMMV), couldn't you create a meetme and play hold music into the meetme and then also play the prompt into the meetme at the same time without interrupting the hold music? This would obviously not work for high load but... JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
Shane D wrote: Try this: exten = 1000,1,Answer() exten = 1000,2,Wait(2) exten = 1000,3,VoiceMailMain() You do not specify the mailbox number in the call to the application. You only specify the number to VoiceMail() HTH, Shane On 1/31/08, Drew Gibson [EMAIL PROTECTED] wrote: John Von Essen wrote: Any ideas what could be going on? I tried tweaking the extension 1000 so it looks like: exten = 1000,3,VoicemailMain,s6000 It may be your syntax, try :- exten = 1000,3,VoicemailMain(6000|s) regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com What do you mean you do not use the mailbox in Voicemailmain see below: *CLI -= Info about application 'VoiceMailMain' =- [Synopsis] Check Voicemail messages [Description] VoiceMailMain([EMAIL PROTECTED]|options]): This application allows the calling party to check voicemail messages. A specific mailbox, and optional corresponding context, may be specified. If a mailbox is not provided, the calling party will be prompted to enter one. If a context is not specified, the 'default' context will be used. Options: p- Consider the mailbox parameter as a prefix to the mailbox that is entered by the caller. g(#) - Use the specified amount of gain when recording a voicemail message. The units are whole-number decibels (dB). s- Skip checking the passcode for the mailbox. a(#) - Skip folder prompt and go directly to folder specified. Defaults to INBOX JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maybe a little OT---USB Handset
Hello All, This may be a little OT for the list but it seems to be to be the place to get the best answer. I have looked at the many Skype/Yahoo phones out there and none seem to be what I am looking for. I have a need for a USB handset that I can use with an Asterisk server. This will be on the server itself and an extension on that server, most likely the only extension. The handset needs the ability to generate its own on hook/off hook and DTMF so that I would not need to load a soft phone. I will eventually be needing many of these so if the set up requires a lot of hacking to the phone it may not be feasible. Having said that any suggestions will be appreciated. I know I could use an ATA and a PSTN Phone from wally world, but this will not fit the project or the need. Thanks, JohnM begin:vcard fn:John Millican n:;John Millican adr:;;PO Box 9;Wentworth;NH;03282;US email;internet:[EMAIL PROTECTED] title:Director of Technology tel;work:603-764-9163 x-mozilla-html:FALSE url:www.sentinelcommunications.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe a little OT---USB Handset
Tzafrir Cohen wrote:
On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked at the many Skype/Yahoo phones out
there and none seem to be what I am looking for.
I have a need for a USB handset that I can use with an Asterisk server.
A USB handset is basically a sound device (and not a great one, usually)
along with a small keyboard. Linux will usually easily identify the
sound device and you can use the phone as chan_{oss,alsa,console}.
Using the keyboard in it may be trickier.
Do any of the above support cancelling acustic echo? Is it actually
needed in this case?
Tzafrir,
Thanks for the reply. Acusitic echo cancel may not be needed as this
will not be used in a noisy work place, only in possibly quieter home
environments. There will also be no need for speaker phone operation.
Enabling the keypad is definitely the tricky part. I am trying to avoid
loading a soft phone since I don't want to have to instruct the users on
how to use one (mostly NON-technical types). If the set looks and feels
like a phone they will be OK on their own. I guess I may have to go
with a decent, hopefully inexpensive, basic IP desk phone.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maybe a little OT---USB Handset
Gordon Henderson wrote:
On Sun, 27 Jan 2008, John Millican wrote:
Tzafrir Cohen wrote:
On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked at the many Skype/Yahoo phones out
there and none seem to be what I am looking for.
I have a need for a USB handset that I can use with an Asterisk server.
A USB handset is basically a sound device (and not a great one, usually)
along with a small keyboard. Linux will usually easily identify the
sound device and you can use the phone as chan_{oss,alsa,console}.
Using the keyboard in it may be trickier.
Do any of the above support cancelling acustic echo? Is it actually
needed in this case?
Tzafrir,
Thanks for the reply. Acusitic echo cancel may not be needed as this
will not be used in a noisy work place, only in possibly quieter home
environments. There will also be no need for speaker phone operation.
Enabling the keypad is definitely the tricky part. I am trying to avoid
loading a soft phone since I don't want to have to instruct the users on
how to use one (mostly NON-technical types). If the set looks and feels
like a phone they will be OK on their own. I guess I may have to go
with a decent, hopefully inexpensive, basic IP desk phone.
I had a little success with a cheap USB 'phone' (From Tesco in the UK)
which was a Yealink device. Linux has a driver for the keypad on it which
makes it work just like a regular keyboard (limited number of keys,
obviously!), but the issue is still that you'd need a program of some
sorts to take the keypad input and translate it to an asterisk console
command dial, if using it as a console phone.
I did use it successfully some time back with idefisk, although idefisk
didn't have a keyboard equivalent of 'hang up' at the time (zoiper might
have now though). The down-side was that you needed to put the mouse over
the idefisk application so it had keyboard input focus )-:
Oh for a command-line IAX client, but it's something I just don't have
time to put together myself.
Gordon
Thanks for the replies.
I wonder if I could use the Yealink phone and write a connector to
Asterisk with the IAX client on Sourceforge and make the handset look
like an iaxphone? Or maybe there is some other easier solution? All I
need is to have the ability to go
off hook/on hook, pass DTMF, and voice obviously :-)
JohnM
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] inbound Audio problems probably not NAT related?
Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108 sleeping, 0 stopped, 0 zombie Cpu0 : 0.2%us, 0.2%sy, 0.0%ni, 97.3%id, 2.2%wa, 0.1%hi, 0.0%si, 0.0%st Cpu1 : 0.3%us, 0.0%sy, 0.0%ni, 99.6%id, 0.1%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 4052276k total, 2586128k used, 1466148k free, 389208k buffers Swap: 4200956k total,0k used, 4200956k free, 1929952k cached from show channels:(was the same before and after top was run) 12 active channels 6 active calls Would any of the guru's here say that this was good, bad, middle of the road, not enough info to tell? At the time I copied this there were 5 active calls in show channels. This server is exhibiting some strange behavior and I was starting to think it may be system overload. I find this hard to accept given the specs but, hey I don't know everything! some info from /proc/cpuinfo: vendor_id : AuthenticAMD cpu family : 15 model : 35 model name : Dual Core AMD Opteron(tm) Processor 180 stepping: 2 cpu MHz : 2411.130 cache size : 1024 KB some info from /proc/meminfo: MemTotal: 4052276 kB MemFree: 1469356 kB Buffers:388196 kB Cached:1927548 kB SwapCached: 0 kB Active: 893644 kB Inactive: 1523168 kB HighTotal: 0 kB HighFree:0 kB LowTotal: 4052276 kB LowFree: 1469356 kB SwapTotal: 4200956 kB SwapFree: 4200956 kB Dirty: 228 kB Writeback: 0 kB Hardware RAID 5 on-motherboard gigE connected through Cisco switch On inbound calls I lose the incoming audio after a couple minutes, outbound audio is always good, then after a while inbound audio magically starts up again. this happens on maybe 10% of calls at its worst. I have looked at the possibility of NAT issues and do not believe that to be the case. I have noticed that the memory usage climbs steadily but I believe that is the kernel as top show no process with more than 0.4% memory usage. Although when I rebooted (yes, an act of desperation) over the weekend the amount of calls with this problem dropped dramatically along with total memory usage which is slowly climbing again. Started at about 1gig on Saturday morning and is now at the 2.6gig shown above in top. This box typically does around 35,000 minutes of calls each month with a couple busy periods each day during weekdays. Normally no more than 10 to 12 calls at one time. provider--T1 to Cisco router--Asterisk--phones The router is doing NAT and routing all traffic from a specific IP to the asterisk box and dropping everything from any other IP. canreinvite is set to no on the sip trunk and all the phones. One thing that may be related is that when I ssh into this box it takes a full minute respond after the pass phrase is typed in. Could this be related or am I just grasping at straws? Any Ideas? -- JohnM begin:vcard fn:John Millican n:;John Millican adr:;;PO Box 9;Wentworth;NH;03282;US email;internet:[EMAIL PROTECTED] title:Director of Technology tel;work:603-764-9163 x-mozilla-html:FALSE url:www.sentinelcommunications.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip calls drop one leg after about 2 minutes
Doug wrote: At 14:54 1/10/2008, John Millican wrote: Hello all, I know this has been discussed before but I am not finding the thread on voip-info or site:lists.digium.com through google. I have a site with * ver. 1.4.15 (started out as 1.4.2 or so) running on openSuSE 10.2, Dual core AMD Opteron, purely SIP. I haver been experiencing a problem where after about 2 min 10 seconds to 2 minutes 18 seconds incoming audio stops. The call is still up but no inbound audio. At first I thought the calls was dropping completely but that is not the case. iirc there was a similar problem a while back in * versions. I looked on voip-info but did not find anything that appeared to be the same issue. This does not happen on all calls, maybe 4 or 5% of calls. I am not finding anything in the log files to tell me what is going on. I am going to upgrade to 1.4.17 tonight and see if there is any difference. Any suggestions of what to look at or where to go (keep it clean ;-) please) would be greatly appreciated. Thanks in advance JohnM NAT? http://www.google.com/search?q=Asterisk+dropped+calls+NAT I don't think it is a NAT problem as the call is established and is great for about 2 minutes then, only the one leg goes away. Nat is being done by a Cisco router (model 1804???), and has all UDP traffic from IP xxx.xxx.xxx.xxx frowarded to the inside asterisk IP address. I have tried nat=yes and externip=xxx.xxx.xxx.xxx in sip.conf but when i do that nothing works. I will try this again over the weekend to confirm that I had it correct. JohnM begin:vcard fn:John Millican n:;John Millican adr:;;PO Box 9;Wentworth;NH;03282;US email;internet:[EMAIL PROTECTED] title:Director of Technology tel;work:603-764-9163 x-mozilla-html:FALSE url:www.sentinelcommunications.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip calls drop one leg after about 2 minutes
Hello all, I know this has been discussed before but I am not finding the thread on voip-info or site:lists.digium.com through google. I have a site with * ver. 1.4.15 (started out as 1.4.2 or so) running on openSuSE 10.2, Dual core AMD Opteron, purely SIP. I haver been experiencing a problem where after about 2 min 10 seconds to 2 minutes 18 seconds incoming audio stops. The call is still up but no inbound audio. At first I thought the calls was dropping completely but that is not the case. iirc there was a similar problem a while back in * versions. I looked on voip-info but did not find anything that appeared to be the same issue. This does not happen on all calls, maybe 4 or 5% of calls. I am not finding anything in the log files to tell me what is going on. I am going to upgrade to 1.4.17 tonight and see if there is any difference. Any suggestions of what to look at or where to go (keep it clean ;-) please) would be greatly appreciated. Thanks in advance JohnM begin:vcard fn:John Millican n:;John Millican adr:;;PO Box 9;Wentworth;NH;03282;US email;internet:[EMAIL PROTECTED] title:Director of Technology tel;work:603-764-9163 x-mozilla-html:FALSE url:www.sentinelcommunications.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] noun-verb vs verb-noun aka dogs black vs black dogs
On Wednesday December 19 2007 6:09 pm, Tzafrir Cohen wrote: On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote: This only works because you are closed to the alternative. The alternative (verb-noun) works fine for the above referenced applications and many more. Do you want to tally the number of users of applications that use noun-verb instead of verb-noun? Is there a reason verb-noun works fine for them and not for us? OK, here's a small usability test to your idea: Here's a partial list of actions from asterisk 1.4. Which of them is supported by your hypothetical MGCP device? (no cheating, please) active add answer audit autoanswer boost clear convert del deltree dial dumphtml flash get hangup logoff mute put reload remove save send set show showkey transfer unmute Okay I have to put my 2 cents in now can't resist any longer even though it may only be worth 0.5 cents. In MY opinion, consistency is first and formost. I can learn almost any command struture IF i put my mind to it and I want to do so. What is hard for me is changing in mid stream. having said that I always liked a drill down structure. Big idea first, followed by category of idea, followed by.. and so on till you get the the exact single item that you are looking for. A US based example: show world north_america us state nh capitol Gives: Concord You could easily do : show world giving all the continents show world north_america giving all countries in North America and so on down the line. To ME and maybe only me, this make since, object world knows of continents, object continents knows of countries, object countries knows of state, object state knows of capitols. Easy for programmers, users and computers alike. again just my opinion. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Fxo card headaches
See Inline On Tuesday December 11 2007 2:20 pm, Jonn R Taylor wrote: The old x100p cards where 5 volt pci cards. I had this same problem and it was the type of pci slot that I had the card plugged into. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Boczko Sent: Tuesday, December 11, 2007 1:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] X100P Fxo card headaches Hello List, Thanks for the replies...currently Still doing the same after adding channel=1 to the /etc/zaptel.conf my zapata.conf looks like orange:~# more /etc/asterisk/zapata.conf [trunkgroups] [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes context=incoming signaling=fxs_ks group=1 channel=1 I believe this should be channel=1 JohnM orange:~# Im getting pretty sure now that its the card, i borrowed a TDM400B (the 3port model) from a friend, and it came up just fine (with a quick change to zaptel.conf and zapata.conf)im thinking. duff card Ive got a linksys SPA-3102 on order to replace this card... but i would still like to confirm if it is the card thats duff or something with my config, more of a geeky pride excercise than anything else. Cheers Chris On 11/12/2007, Drew Gibson [EMAIL PROTECTED] wrote: Chris Boczko wrote: Hello List, Im just dipping my feet into the asterisk world, and im having major fxo problems Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons (from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a Debian Etch box, with 1gb ram, running all of the services for my home server (web / db / music server etc), and i would like to run my PSTN line from Kingston Comms, but i can't get this box the recoginsie this line! The X100p is a cheap clone i got off ebay for a tenner, so im not expecting much, i know they have echo issues, but im going to upgrade to a SPA3012 / TDM400B when i have the cash. Ztcfg -vv reports orange:~# ztcfg -vv Zaptel Version: SVN-branch-1.4-r3374 Echo Canceller: MG2 Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels to configure. orange:~# and my zaptel.conf file contains orange:~# more /etc/zaptel.conf fxsks=1 loadzone=uk defaultzone=uk orange:~# but zap show status in the command line shows orange*CLI zap show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard X100P Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) orange*CLI and i think the Unk means unknown...? Does anyone have any ideas on how to get this line to work, ive followed every howto i can find, and google seems to be comming up short, as far as i can see, ztcfg should report the card as configured, but it isn't, and ive no idea why. Hope you can help Chris Try adding channels=1 to the end of your zaptel.conf to assign the settings you have made to a channel. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip 1.4.x DTMF detection not working
Hello
I have a setup where i have 2 asterisk servers connected over the public
internet with plenty of bandwidth, NAT on one side only. If i use IAX
between the two *'s dtmf is flawless. If I use SIP, DTMF detection is around
30% or less. I have an exten to dial into and check DTMF:
exten = NPANXX,1,Answer(); (actual number blanked for privacy)
exten = NPANXX,n,Read(userChoice|ogm/intro|4||1|4);
exten = NPANXX,n, SayDigits(${userChoice});
exten = NPANXX,n,Hangup();
When i dial in and use IAX between the servers i always get all 4 digits, If I
dial in using SIP between the two servers with dtmfmode=rfc2833 or
dtmfmode=inband I MIGHT get 1 or 2 digits. If i use dtmf=info and I dial
slowly I usually get 4 correct digits, but not consistently enough to call it
good, maybe 85%. If I dial 1 2 3 4 quickly I get 1122 or 1223 or the like.
I would like to use SIP as the voice quality seems to be better, matter of
opinion I am sure but...
Both Asterisk's are 1.4.x on SUSE 10.2 x86_64 kernel 2.6.18.2-34
AMD opteron Dual-Core AMD Opteron(tm) Processor 2212
and
Dual Core AMD Opteron(tm) Processor 180
2GIG memory
I have searched voip-info and google and didn't find anything that looked
relevant, maybe just my search words. I do seem to remember something on the
list about this a couple months ago but I can not find it or I am remembering
incorrectly.
Any suggestions will be greatly appreciated.
Thank You,
JohnM
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Speakerphone
On Monday November 12 2007 9:38 am, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric I have found the polcom speaker phone to be very good on the 320's, 330's, and the 501's. Clear clean voice even in relatively noisy areas. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Speakerphone
On Monday November 12 2007 1:50 pm, Doug wrote: At 08:38 11/12/2007, Eric Jacksch wrote: Hello all, We're using a lot of the linksys phones, and while user feedback is generally positive, the speakerphone leaves a bit to be desired. For those of you using the polycom desk phones, how do you find the built-in speakerphone? Thanks, Eric Excellent speakerphone. Extremely cumbersome to configure. I do not understand how you can say that the Polycoms are Extremely cumbersome to configure. I find them rather nice. Once you have one working config it is very easy to copy that config over to the mac address files for the other phones that you have and only change the per phone bits. Set up a site wide sip.cfg and then use phone-(macaddress).cfg files for the individual settings for each phone. real nice when you have more than a couple phones to configure. It is not my intention to start any war here just giving my 2 cents worth. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ABE, Sangoma, T-1 no recognizing calls
Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a Shark Box which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back and swapped cards in the shark and now it is a T-1 out. Wanrouter, zaptel and asterisk are all apparently happy. When I place a call to * I hear ring on the calling side but do not ever see anything in happen on the * side. When I try to call out i get: Executing Dial(SIP/xxx.xxx.xxx.xxx-ab5012d0, zap/3/603xxx) in new stack -- Called 3/603xxx And nothing else, at one time I was getting a zap/answered line but no more. Relevent zapata.conf [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A101 port 1 [slot:8 bus:3 span: 1] context=from-pstn group=0 signalling=fxo_ls channel = 1-6 zaptel.conf loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:8 bus:3 span: 1] span=1,1,0,esf,b8zs fxols=1-6 Extensions.conf [from-pstn] exten = _X.,1,Dial(zap/3/603xxx); Very simple setup at this moment, nothing fancy. I am able to dial in via sip and asterisk answers and send the call to the from-pstn context at which point i see Executing Dial(blah, blah) in new stack; I believe at this time that the problem is in the setup of the shark box. Verizon tells me that there end is good and the T-1 is esf, B8ZS, loop start. But I thought I would ask the list for some opinions before I started pointing the finger. Thank you for any help JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Video Conference
Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am considering at this point is hacking up my own solution using off the shelf equipment. Decent web camera, Polycom conference phone(maybe if the budget holds) and a large wide screen LCD monitor all connected to * Sound reasonable or am I living a pipe dream? JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Conference
sniped and moved to below for readability John Millican wrote: Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am considering at this point is hacking up my own solution using off the shelf equipment. Decent web camera, Polycom conference phone(maybe if the budget holds) and a large wide screen LCD monitor all connected to * Sound reasonable or am I living a pipe dream? JohnM On Monday October 22 2007 9:32 am, SIP wrote: Direct single line video conferencing via SIP is actually pretty straightforward and works rather well. Multipoint conferencing is where you get into a bit of a mess. There are precious few products out there that claim multipoint SIP video conferencing capability, and we've had no luck so far with any of it being what one might consider straightforward. N. Thanks for the responce. Have you had any luck at all even with what one might not consider straight forward? I am trying to avoid paying the $1000+ per location needed to purchase something from say Polycom or Tandberg. I would even be willing to do something along the lines of a web app for video and some how tie that together with the voice through Asterisk. Just don't want to look like one of the old dubbed over Japanese movies from when I was a kid (lips move and then a couple seconds later you hear voice). JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote: Steve Edwards wrote: On Sat, 13 Oct 2007, Lee Jenkins wrote: I have been using axVoice.com for some about 9 month to a year now and their service is pretty damn good. For home users they have unlimited plan for around 22.00-24.00 U.S. per month. I think the pay as you go plans make more sense for most people -- why do you think the vendors push the flat rate plans? At $25.00 per month, you'd have to be on the phone for about an hour a day for it to be cheaper than a $0.015 per minute plan. True, but I work from home, have a wife and 4 kids with friends and family all over the U.S. so it makes more sense for me. Good point though, Steve. --- Lee Be sure to read the fine print as most of the unlimited plans do actually have a limit on usage (even the ones I offer). Some are out in the open some are very well hidden and some others do not even publish the number of minutes that will get you moved to a business rate or possibly even canceled. Think of it from a business perspective, You would not want your clients to use so many minutes that it ended up costing you money, would you? So what the provider has to do is settle on an average of all the customers so that some can use a few more minutes than they pay for and some use less than they pay for, the group in the middle make up the profits. Even under this scenerio there is still a point where the provider starts to loose money. This is why I usually guide customers to a per minute rate so that it is fairer to both sides. Everybody knows what the rules are. Then again there are those that like the convenience of writting the same check every month. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Affordable SIP Trunk for Home PBX ?
On Saturday October 13 2007 12:47 pm, Doug Lytle wrote: John Millican wrote: On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote: Be sure to read the fine print as most of the unlimited plans do actually have a limit on usage (even the ones I offer). Some are out in the open some Then don't advertise it as *unlimited* Seems simple, doesn't it? Doug Doug, You are absolutely correct, it should be simple but... When you are trying to market a product and are competing in a market littered with limited unlimited plans and knowing that these are the key words that a lot of people look for, the other partner in the company said we have to have an unlimited plan. Long hard battle ensued but we came to an agreement. Okay, but we will call it the Unlimited3000 (yes pretty cheesy I know) plan and spell it out clearly in the contract also. Also if you left the rest of the post in you would see that I said I usually steer people away from these plans into a per minute as most people do not get close to 3000 minutes a month. Yes, there are those that do but not the majority. My apologies if anyone feels this is to close to a commercial post but I felt I should answer. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Wednesday October 10 2007 2:15 pm, Doug Lytle wrote: Russell Bryant wrote: I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. What is your opinion? I certainly want the release naming to be as obvious as possible. I would say the rc-1, rc-2 is about as obvious as it gets and would get my vote. JohnM -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 Phone (603) 764-9163 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sine Dialer, GNU dialer, VICIDial and others slightly OT?
Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of integration with asterisk, stability, and such. Thank You for any help JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound dialing
On Wednesday August 08 2007 8:28 am, Tim Johnson wrote:
Hi Drew. Thanks for the tips. My Line 1 works as I'd like it to, and I
could be wrong, but I don't think changing the dialplan there will help. I
really just want to be able to dial local phone calls (7 digits) and have
it go out the SPA3102, without having to dial twice. This is a snip what I
have so far.
extentions.conf
exten = _NXX,1,Dial(SIP/201/${EXTEN},20)
exten = _NXX,2,Hangup
sip.conf
[201]
type=friend
username=x
secret=x
host=dynamic
context=sip
nat=yes
canreinvite=yes
qualify=yes
subscribecontext=localextensions
dtmfmode=rfc2833
vmexten=voicemail
disallow=all
allow=ulaw
allow=gsm
On the SPA (in the PSTN Line tab)
Dial Plan 1: (xxxS0:@gw0)
Dial Plan 2: S0:255
DialPlan 1 is just what I have for now
DialPlan 2 is my extention 255, with a PSTN call comes in it rings my SIP
phone.
I set my VoIP Answer Delay to 0 (from 1, I also tried 5 some time ago) and
I set the SPA To PSTN Gain to 5 and now 15.
With things the way I have them now, when I dial a local number, I get a
single DTMF tone on the phoneline, not sure what digit it is.
snip
I believe you want to put the dial plan in line 1 and reference gw0 there.
On the SPA (in the PSTN Line tab)
Should be in line 1 tab
Dial Plan 1: (xxxS0:@gw0)
Dial plan should be in line 1 dial plan:
normal-dial-plan| xxx:@gw0|[49]11:@gw0|some-more-dial-plan-if-needed
notice where the is in relation to the digits
this will send all seven digit calls out the PSTN and also all 411 and 911
calls out the PSTN line and also 411 and 911 calls. If you leave the PSTN
dial plan as factory default it should work.
If memory serves.
JohnM
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wednesday August 08 2007 12:10 pm, Mike wrote:
I can be a bit slow sometimes, but you said that it's not possible, and on
the other hand told me to write my own function (which appears to
contradict the first statement).
Your example of the use of a function is exactly what I need (Create a
function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})) , what I don't know is
how to actually write the function with a return value (and Googling this
doesn't get me any relevant result, apparently).
I'd be most thankful for some link to a page that shows how to write such a
function in Asterisk.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 08, 2007 11:59
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to write a function with a return
valueinAsterisk
On Wednesday 08 August 2007 11:41:38 am Mike wrote:
But what if I wanted to write my own custom application for one
specific purpose, I can't set a return value? It's not possible at all?
Not possible, to my knowledge.
Let me put it this way then, if I needed to have some processing all
done in the same Asterisk priority (in my case, I want to use the hint
priority but I need to find the value of a variable and use it in the
same line).
Create a function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})
Exten = 12345,hint,Dial(SIP/${A}) ; I need to know ${A} first, but I
can't know before this line is called (it's very DB driven).
Give the function method a try; that's about the only way I can think of
doing something like that... Note that if it's a very DB driven system,
you can use func_odbc to do what you want by declaring an SQL statement as
a function.
-A.
Asterisk will listen on stdin if you have your agi code write the var and
value out to stdout asterisk will then be able touse that var in the dial
plan. this is how I do this in a C++ app that i use often:
fprintf(stdout,EXEC SETVAR RESERVED=1 \n);
then in the dial plan I look at the value of ${RESERVED} and use a gotif to do
what needs to be done based on that value.
JohnM
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 320 - Can it actually be configured?
On Wednesday August 01 2007 5:49 pm, Douglas Garstang wrote: Don't know about the 320, but we provisioned the 301's. They're provisioning is basically the same as the 501's and 601's. What problems are you having? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Sent: Wednesday, August 01, 2007 2:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 320 - Can it actually be configured? Just got one of these. Horrible to program. Trying to key in the FTP server. Won't even remember the info after rebooting. Anybody know the proper way to beat on this stupid beast so it will work? They provision exactly as do the 501, 330, 601 and such. Search voip-info.org and you will find several nice documents on how to do this. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
On Tuesday July 31 2007 4:44 pm, Joe acquisto wrote: . . . Even if you can find non-original-artist recordings of such music, the *compositions* are registered with BMI and ASCAP, and you'll need blanket licenses to play them. (Well, if you only wanted one or two tracks, you might negotiate specific licenses, but I'm not sure it would be cheaper.) Cheers, -- jra So, if, for instance, someone were to pipe in some broadcast stations, for MOH, that would be a copyright violation? Not that I know how to do that, with *, off the top of my head. joe a IANAL or even close to one. Just grabbing the music form a broadcast station and rebroadcasting is technically illegal, at least in the US. What I have been told is that you would first have to check with the broadcast station, sign an agreement with them, and depending on the scope of their coverage, as in geographic, you might be able to use their station. It is my understanding that radio stations pay a license fee based on the coverage/market area. You may have to pay them the difference between what they pay for the coverage of the station and the global coverage that a PBX could potentially have. As a side note there have been several examples posted in the past of how to pipe in a music source. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP 4000 Soundstation SIP Conference Phone Question
On Monday July 23 2007 9:26 am, Matt wrote: Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound? I have one at a customer site and they are very happy with it. Works well, sound quality is good, typical Polycom speaker phone sound. You may have to train user not to yell though. JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP NTP Clock skew
Hello All, I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34 I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been getting: Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136, dlsr=196500 (2:998ms), diff=664 I see an entry in Mantis that Russell fixed code so that this will not show when it shouldn't. Would i be correct in assuming that if i pull a copy of 1.4.5 from digium this weekend that this message will go away? Also, just to show off my ignorance, what is this message telling me? Is this simply a deference between unix time-t and NTP timestamps and therefore nothing of much concern? JohnM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTCP NTP Clock skew
On Thursday June 28 2007 1:19 pm, Jared Smith wrote: On 6/28/07, John Millican [EMAIL PROTECTED] wrote: Would i be correct in assuming that if i pull a copy of 1.4.5 from digium this weekend that this message will go away? No... you'd have to pull the latest code from the 1.4 branch using Subversion, or wait for 1.4.6 to be released. -Jared Thank you for the info. I missinterpreted Russell's comments to by that it would be in the 1.4 stable, silly me. Internal RTCP NTP clock skew detected: lsr=2362715969, now=2362741181, dlsr=65500 (0:999ms), diff=40288 So what sort of badness will this be causing, if it is not fixed in 1.4, by me waiting until 1.6? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA941
On Thursday June 14 2007 1:12 pm, Matt wrote: We had several of these when we were first playing around with Asterisk. They are somewhat nice. The audio quality left some to be desired, however, we did not have a hold button issue. On 6/14/07, Shad Mortazavi [EMAIL PROTECTED] wrote: Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi I just installed 64 of these for a customer and the hold works on the ones that have been tested. We are not on the latest firmware yet though. I will be testing that tomorrow. John M -- John Millican Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 Phone (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DISABLE 9?
On Sunday April 15 2007 5:48 am, JNA wrote: Is there a way to make it so you do not have to dial 9 by default to dial a outside number? I would like it if we could just dial the number any pointers? In a number of my ATA's and IP Phones I have a delay in the pattern match so that if the user dials 4 digits the phone waits for 1 second to see if there will be a 5th or more digits. This eliminates the need to dial a 9 or a 0 to get outside dial tone. Yes 1 second can be a long time but if you don't make a big deal over it when talking to the users they usually do not notice and are more focused on the fact that they do not have to do anything to distinguish between extension dial and outside dial. John M ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA 3102
On Thursday October 12 2006 4:15 pm, Dave Cotton wrote: On Thu, 2006-10-12 at 21:30 +0200, Csibra Gergo wrote: Thursday, October 12, 2006, 6:58:57 PM, Tim wrote: I've read alot of comments on the SPA-3000, many if not all saying they had echo issues, but I've not seen anyone comment on the SPA-3102. Does anyone have any comments or issues with these? Well, I have had echo issues. Then I find out the echo cancellation on PSTN line is switched off by default. I switched on, and no echo any more :) I have had echo with the SPA3000 but I switched to Global impedance on the FXO and since then clear as a bell. I have several of these in the field with residential users and no complaints. -- John M ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Home Hardware SIP Proxy with use of POTS in Emergency
On Monday October 09 2006 6:53 pm, Brandon Galbraith wrote: Does anyone know of any ATA devices (Linksys, Dlink, Cisco, etc) that will fail over to POTS for an emergency call? I'd like to route any call except a 911 call over SIP or IAX, but any 911 call should be routed out over POTS. If this is not an option, I'm also open to devices that will fail over to GSM to make the emergency call. I apologize if this topic has already been covered before. -brandon Sipura 3000 or 3102 to start with I am sure there are others -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 Phone (603) 764-9163 fax (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] balance anouncement
On Friday September 01 2006 9:27 am, ram wrote: Hi how can i do balance anouncement by using asterisk take example, i have table balance , user name 9, balance 200$ user dial *98 or what ever, then i need anouce his balance is 200$, by reading from that row any clues how can i achive this or is this possible ? Ram Create an AGI script that does a db look up for the ballance and then pass the result back to Cepstral or Festival or your favorite text to speech software. John M ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] balance anouncement
On Friday September 01 2006 10:19 am, ram wrote: Hi thanks for the quick reply any documents to read to achive this or any examples would be great to read Ram On 9/1/06, John Millican [EMAIL PROTECTED] wrote: On Friday September 01 2006 9:27 am, ram wrote: Hi how can i do balance anouncement by using asterisk take example, i have table balance , user name 9, balance 200$ user dial *98 or what ever, then i need anouce his balance is 200$, by reading from that row any clues how can i achive this or is this possible ? Ram Create an AGI script that does a db look up for the ballance and then pass the result back to Cepstral or Festival or your favorite text to speech software. John M Try google or voip-info.org and search for Asterisk AGI should yeid some good results. AGI can be called from the dial plan and written in your favorite language i.e. PHP, C++, Perl, C, Java or start here: http://home.cogeco.ca/~camstuff/agi.html http://asterisk.drunkcoder.com/agi.cgi John M ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can not get ${LEN(VAR)} and greater than to work for me
On Saturday August 26 2006 11:15 am, Matt Riddell (IT) wrote:
John Millican wrote:
Hello all,
I am trying to test if the length of a dialed number is greater than 7.
When i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial) in
new stack
indicating that the number was not greater than 7.
if i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}=11]?dialout:nodial);
and dial the same 1 800 xxx
i get:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 1?dialout:nodial) in
new stack
indicating that the length of number dialed was equal to 11 digits.
so equal to works and greater than does not?
Can any one see what I am doing wrong?
* version 1.2.9.1
Maybe string comparison because of the speech marks?
Thank You Matt and Ira
The speech marks/quotes were the problem.
Matt sorry about the earlier direct mail used R instead of L for the reply.
John M
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can not get ${LEN(VAR)} and greater than to work for me
Hello all,
I am trying to test if the length of a dialed number is greater than 7. When
i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial) in new
stack
indicating that the number was not greater than 7.
if i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}=11]?dialout:nodial);
and dial the same 1 800 xxx
i get:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 1?dialout:nodial) in new
stack
indicating that the length of number dialed was equal to 11 digits.
so equal to works and greater than does not?
Can any one see what I am doing wrong?
* version 1.2.9.1
TIA
John M
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 dialplan strings.
On Tuesday August 22 2006 7:24 am, Ken D'Ambrosio wrote: I'm trying to set up a dialplan that dials via PSTN for: All eight-digit calls that start with 9 All 911 calls All calls that start with 424 (the local exchange) I haven't tested 911 -- for obvious reasons. I may do so after I feel more confident. I've got the starts-with-9 working fine. But the local exchange stuff isn't working, and I'm confused. Here's a snippet of my dialplan: [lots deleted]|9,:xxx :@gw0|424 :@gw0) It does dial 424 numbers, but they go straight through SIP. Any suggestions? Thanks! -Ken Ken, Just a hunch but it may be the space in the dial string between the and the : Your string: 9,:xxx :@gw0|424 :@gw0) corrected: 9,:xxx:@gw0|424:@gw0) as I said just a guess. -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and Asterisk
On Wednesday August 16 2006 7:01 pm, Don wrote: Has anyone used Cepstral for text to speech before? I am testing the demo and it seems to take about 20 seconds for the speech to start... On a 3.4Ghz 2GB machine... Thanks, Don Don, I have been using Cepstral for about a year now and it has worked very well. It starts speaking almost immediately. You definitely know it is a computer voice but thats okay for my application. The following is swift.agi that I found on voip-info.org (I cant remember the author or I would credit him here, my apologies.) The last line documents how to use. ## #!/bin/sh #Assign the value sent from the exten= line to $text so it can be used below text=`echo $*` #Set $stdin to something stdin=0 while [ $stdin != ] do read stdin if [ $stdin != ] then stdin2=`echo $stdin | sed -e 's/: /=/' -e 's///g' -e 's/$//' -e 's/=/=/'` eval `echo $stdin2` fi done calleridnum=`echo $agi_callerid | cut -f2 -d\ | cut -f1 -d\` calleridname=`echo $agi_callerid | cut -f1 -d\ ` /opt/swift/bin/swift -o /tmp/$agi_uniqueid.wav -p audio/channels=1,audio/sampling-rate=8000 $text #Now, tell asterisk to play that file echo stream file /tmp/$agi_uniqueid # #Read the reply from asterisk to our command read stream #Clean up our mess and delete that file rm /tmp/$agi_uniqueid.wav exit 0 # exten= s,1,agi(swift.agi|This is some text\, which needs to be converted to speech.) ## I have used this (on a very low call volume obviously) on as low end a machine as PII 400 with 512 meg ram. Hope this helps -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asking for phone number to dial
Instead of Background() use Read(). this will allow for any number of digits.
example:
exten = 1234, 1, Read(var_to_use|prompt_name|number_of_digits_to_accept);
;then use a goto based on the value of var_to_use.
exten = 1234,2 GoToIf($[${var_to_use} = 1]?new_exten,1:3);
this way you are sent to an extension based on what the user dials. this
will fall through till it matches.
John M
On Friday June 23 2006 6:18 pm, Anthony Cennami wrote:
Where there is a will, there's a way:
Assuming you only had one DID, you could:
- Use an auto attendant
- Use a dialplan timeout that dropped to DISA (not so nice)
- Use callerid based redirection (again, not so nice, but available)
- Use a Voicemail breakout option (again, not so nice, but better than
timeout)
- Use a ToD based extension
Probably a variety of other options, depending on the
application/requirements/costs/etc.
On 23 Jun 2006 22:08:20 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I thought Background() only allowed you one digit dialing while it's
playing.
Is this not the case? I agree with the reply which said that you want to
use DISA, the only problem with DISA is that you have no way to use the
line
for answering regular calls. Once you put the DISA command in the
dialplan,
you get the DISA dialtone for entering you code. I suppose if you know
where
you will be calling from, you could code in a specific dialplan based on
your
callerid info, but that just seems kind of tedious just for being about
to dial out.
Undrhil
--- Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com wrote:
The number dialed after Background
is stored in the EXTEN variable and can be used in the Dial application.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Don
Sent:
Friday, June 23, 2006 5:29 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Asking for phone number to dial
background just accepts input while other sounds...etc...are being
played...
instead of waiting for something to end and then accept input.
It doesn't store the number...etc...then add it to dial command for a
zap
channel
- Original Message -
From: T. Shaw [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, June 23, 2006 5:19
PM
Subject: RE: [Asterisk-Users] Asking for phone number to dial
Isn't that what the Background() application does?
[EMAIL PROTECTED]
blah...
From:
Don [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List -
Non-Commercial
Discussionasterisk-users@lists.digium.com
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asking for
phone number to dial
Date: Fri, 23 Jun 2006 15:51:00 -0400
Does
anyone know where to find an example or able to provide an example of
how to do the following:
When asterisk answers a call...
Ask
for number to dial...then dial that number?
I am basically dialing into
the asterisk box and then wanting it to take
the digits I enter and
dial them on an outbound zap trunk...
I basically am just not sure
how to have asterisk accept the digits and
then use them in the dial
command...
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth
and Colocation provided by Easynews.com --
Asterisk-Users mailing
list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
No virus found in this incoming message.
Checked
by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.9.2/373 -
Release Date: 6/22/2006
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation
provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided
Re: [Asterisk-Users] DTMF Talk off
Okay here goes,
I guess I misunderstood Doug's question about the far end interface. I have no
availability for high speed internet at my house to place a VoIP call over.
So, I have a standard phone plugged into the PAP2, The PAP2 plugs into the
network at my house to which the asterisk box is also connected, the
asterisk box has an FXO card that has the PSTN line plugged into it, this is
where the ZAP channel comes in. when i dial a local number asterisk simple
dials the number out the pstn line. If i dial a long distance number, the *
box dials a local phone number that I have through my VoIP provider which is
answered by an * box that I have at a different location using a line in
extensions.conf like:
Dial(zap/1/my_sip_numberww${EXTEN});
this way when the second * answers the phone it get the ${EXTEN} that I
actually dialed and dials it out over the cable connection. I hope i was a
little clearer this time and sorry for the confusion.
John M
On Monday June 19 2006 11:22 pm, Mike Fedyk wrote:
this does not make any sense.
How do you dial to a service provider from your * box? Does it use PPP
and IP? And then you connect to another * box that is on a cable
connection that receives the call over IP and then dials out to a voip
provider? How do any fxo devices come into this picture? How does a
zap channel come into this picture?
John Millican wrote:
Doug,
The interface that i dial to is at my Service provider and am not sure
what they are using. I dial out of my * box to a service provider number
which is answerd by an asterisk box that I have at another location on a
high speed cable connection, that box then dials the numberI ultimately
want to reach. I use an extensions.conf line at my home * such as:
Dial(zap/1/my_sip_numberww${EXTEN});
this works great and saves me a ton on call costs.
John
On Monday June 19 2006 12:19 pm, Doug Crompton wrote:
John,
You said you were using a PAP2.. what is the FXO interface at the (far)
asterisk end?
Doug
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307 *
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
I have looked at that as a solution but haven't been able to get the dtmf to
work reliably. When I dial a local call i get connected okay and can
obviously connect to the second * box with out problem, the problem comes in
trying to get the ${EXTEN} portion of the dial string
Dial(zap/1/my_sip_numberww${EXTEN});
to the second * box. It tends to not see the DTMF correctly for the number I
want to call. When I watch this on the CLI through SSH on the second box i
see the call come in and go to the correct extension where it waits for
digits to dial and I get sporadic results, sometimes no digits are recognized
and sometimes 2 or 3, but never all correctly. I have tried increasing, and
decreasing the wait in the dial string in my home * with no luck. Any hints
on how to get the 3000 and the * box to talk better? If I could get this to
work through the 3000 I would be a very happy camper as it would open up some
possibilities that I can't do cost effectively otherwise. I will start to
route all local calls out the 3000 though for testing in the mean time.
Thanks for the ideas,
John M
On Tuesday June 20 2006 10:16 am, Doug Crompton wrote:
Ok Now I understand. You mentioned you have an SPA-3000 in your inventory.
That is what I use here and I do not load or use zap or pri modules. I use
the 3000 as my fxo/fxs via sip on my local network. I have no cards in my
computer. You could do the same for testing of your problem.
Doug
On Tue, 20 Jun 2006, John Millican wrote:
Okay here goes,
I guess I misunderstood Doug's question about the far end interface. I
have no availability for high speed internet at my house to place a VoIP
call over. So, I have a standard phone plugged into the PAP2, The PAP2
plugs into the network at my house to which the asterisk box is also
connected, the asterisk box has an FXO card that has the PSTN line
plugged into it, this is where the ZAP channel comes in. when i dial a
local number asterisk simple dials the number out the pstn line. If i
dial a long distance number, the * box dials a local phone number that I
have through my VoIP provider which is answered by an * box that I have
at a different location using a line in extensions.conf like:
Dial(zap/1/my_sip_numberww${EXTEN});
this way when the second * answers the phone it get the ${EXTEN} that I
actually dialed and dials it out over the cable connection. I hope i was
a little clearer this time and sorry for the confusion.
John M
* Doug Crompton *
* Richboro, PA 18954*
* 215-431-6307 *
**
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to use a data T-1?
Warren, My suggestion for testing would be just use ethernet hand off to the asterisk from the Cisco. You could bypass the Cisco but then you would need a T-1 card for the asterisk box and they are not cheap. I believe there are valid arguments for both choices though and ultimately should be decided by what you are planning as a final solution. John M On Monday June 19 2006 10:15 am, Warren wrote: I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to use a data T-1?
Warren, Yes. The setup is based on what type of signaling the telco is giving you. John On Monday June 19 2006 10:32 am, Warren wrote: John, Thanks for the quick reply. I do intend to get a T-1 card anyway. Would it be the same card for a data T-1 as for a voice T-1 just with different setup? W John Millican wrote: Warren, My suggestion for testing would be just use ethernet hand off to the asterisk from the Cisco. You could bypass the Cisco but then you would need a T-1 card for the asterisk box and they are not cheap. I believe there are valid arguments for both choices though and ultimately should be decided by what you are planning as a final solution. John M On Monday June 19 2006 10:15 am, Warren wrote: I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
Doug,
thanks for the help. I am using uLAW and inband every where. I have tried
using 2833 and it did not appear to make any difference, better or worse.
this is why I was thinking that if I could increase the minimum required time
for a tone that it night help, I am just not sure where the best place top do
this is. i thought I had seen a post about setting relaxdtmf to a value to
actually make dtmf detection stricter but i can not seam to find anything
other than 'yes' or 'no'.
John
Doug Crompton wrote:
John,
Well I am certainly not an expert on this. I am using an SPA-3000 and I
have not experienced this. I did have to go to inband on the fxo channel
as rfc8322 did not work for ivr's when using Asterisk. I think you said
you were using a linksys or sipura product for you fxo?? If that is the
case using inband and the ulaw/alaw encoder for the fxo channel might
help. Worth a try I guess. There are some rfc8322 issues that apparently
will be addressed with a rewrite in the next makor version release.
Doug
On Mon, 19 Jun 2006, John Millican wrote:
Doug, I read that post and unfortunately it was not a solution. I do not
believe it has to do with interstate as it happens intra state also. Is
there any way to make DTMF detection stricter, ie require a longer
minimum tone length. Assuming ( yes a dangerous practice) that the human
voice will not hold a DTMF sequence stable for very long, if I lengthen
the minimum required length I may be able to minimize the talk off. What
do you think? Any suggestions?
John M
Doug Crompton wrote:
Check
http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.ht
ml
On Sun, 18 Jun 2006, John Millican wrote:
Hello all,
I have seen some chatter again about DTMF. I see most of the talk
about DTMF around not being able to get an external IVR to recognize
digits, not a big issue for me at this time but sill interesting. My
issue though, is with talk off on a zap channel. It seems to be
getting worse or maybe my patience is thinning. All my calls go out
and come in pstn through an FXO as I do not have high speed available
here at home. My Current setup is:
Phone--PAP2-- * ---PSTN---Voip number to * at another
location(that has high speed)---send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I
also remeber seing something about relaxdtmf being set to something
other than yes or no, so I looked in chan_zap.c and found relaxdtmf
in many places but it looked to my inexperienced eye that it could
only be set to 'yes' or 'no', but i did find some mention of
tonlength (at line 10858) followed that to zaptel.c (line 3357) where
it said :
if ((tdp.dtmf_tonelen 4000 ) || (tdp.dtmf_tonelen 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values
ignore it. Any ideas what might happen if i increased the minimum
time value? if this is indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel = 1
same for channel 2
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten = s,1, NoOp(${CALLERID} time ${DATETIME});
exten = s,2, Dial(sip/677sip/666,30,tT);
exten = a bunch of stuff to do with agi look ups and voicemail
leave/retrieve
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954*
* 215-431-6307 *
**
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
Doug,
The interface that i dial to is at my Service provider and am not sure what
they are using. I dial out of my * box to a service provider number which is
answerd by an asterisk box that I have at another location on a high speed
cable connection, that box then dials the numberI ultimately want to reach.
I use an extensions.conf line at my home * such as:
Dial(zap/1/my_sip_numberww${EXTEN});
this works great and saves me a ton on call costs.
John
On Monday June 19 2006 12:19 pm, Doug Crompton wrote:
John,
You said you were using a PAP2.. what is the FXO interface at the (far)
asterisk end?
Doug
* Doug Crompton *
* Richboro, PA 18954*
* 215-431-6307 *
**
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] home routers
Shaun, I believe that there are 2 models of the WRT54GP2 as there was/is with the PAP2's one that is set for VONAGE and one that is not, typically referred to as the WRT54GP2-NA John M On Monday June 19 2006 3:37 pm, Shaun wrote: I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: DTMF Talk off
Matt, Thank you very much! I am currently running 1.2.7.1 but will be upgrading to 1.2.9.1 this week. I will try toneduration=200 first and let you/list know how well it worked. I read in zapata.conf.sample where it says: How long generated tones (DTMF and MF) will be played on the channel (in milliseconds) and did not realize that would have an effect on recognition. Thanks again, John M On Monday June 19 2006 2:58 pm, Matt King wrote: With recent versions of *, you can increase the detection time in zapata.conf with the toneduration variable. The default setting is: toneduration=100 We had the same problem and solved it by increasing this to 200. You can also increase the threshold volume for detection of DTMF by setting VPM_DEFAULT_DTMFTHRESHOLD in the relevant zaptel wctX.c and recompiling (though if you increase this too much you risk losing your ability to detect DTMF at all). Hope this helps, Matt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
Doug,
The PAP2 is single ethernet and 2 fxs. I actually have a couple of these a
SPA-3000, and a SPA 2102 (testing purposes). these connect to asterisk over
the network and * is the house PBX set to dial a local Voip Number via the
zap/PSTN, which is routed to another * box on a cable connection that dials
out over the cable back to the VoIP Provider and then routed to the world.
this way i can still have the lower cost of VoIP while living in my back
woods New Hampshire location where High speed is not available. I currently
have three PSTN lines here at the house, 1 line for dial-up and 2 plain
vanilla local only lines.
John
On Monday June 19 2006 5:28 pm, Doug Crompton wrote:
Is the PAP2 an ethernet connected device to * ? I was wondering why you
were using zap if it were not an internal card?
Doug
On Mon, 19 Jun 2006, John Millican wrote:
Doug,
The interface that i dial to is at my Service provider and am not sure
what they are using. I dial out of my * box to a service provider number
which is answerd by an asterisk box that I have at another location on a
high speed cable connection, that box then dials the numberI ultimately
want to reach. I use an extensions.conf line at my home * such as:
Dial(zap/1/my_sip_numberww${EXTEN});
this works great and saves me a ton on call costs.
John
On Monday June 19 2006 12:19 pm, Doug Crompton wrote:
John,
You said you were using a PAP2.. what is the FXO interface at the
(far) asterisk end?
Doug
* Doug Crompton *
* Richboro, PA 18954*
* 215-431-6307 *
**
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954*
* 215-431-6307 *
**
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
