[asterisk-users] doing dnsmgr_lookup
Hello list, my CLI is spammed with : [Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] doing dnsmgr_lookup for 'ssw4.itsp.tld' How can I turn this off ?! dnsmgr.conf : [general] enable=no; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200; refresh managed DNS lookups every n seconds ; default is 300 (5 minutes) Kind regards, Jonas. -- _ -- 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] Moving from DSL to T1
On Sun, 12 Sep 2010, Kevin Keane wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Sunday, September 12, 2010 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving from DSL to T1 On Sun, 12 Sep 2010, Kevin Keane wrote: What really matters is the latency, and T-1 is a huge improvement over DSL in that area. The easiest way to measure latency is the ping time to a server that is ?close to you? Internet-wise. A DSL has latencies of between 40ms (if it?s extremely good and not too many other people are using it) and 1000ms (if there is a problem somewhere). A good T-1 may have latencies as low as 5 ms or so. Also, with a T-1 the bandwidth is guaranteed and bidirectional. With a DSL line, you may get burstable bandwidth ? you don?t actually have that bandwidth, you just get to compete for excess bandwidth with your neighbors. You are confusing DSL with cable. Both, actually. The latency numbers are actual numbers measured at a customer site. Ouch. My home/office ADSL connection (8Mb in/832Kb out) - ping to my external default route: $ ping -q -c10 188.39.1.26 PING 188.39.1.26 (188.39.1.26) 56(84) bytes of data. --- 188.39.1.26 ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9008ms rtt min/avg/max/mdev = 12.913/13.266/13.747/0.286 ms Ping to one of my hosted servers: $ ping -q -c10 unicorn.drogon.net PING unicorn.drogon.net (195.10.225.68) 56(84) bytes of data. --- unicorn.drogon.net ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9009ms rtt min/avg/max/mdev = 17.801/18.443/19.994/0.610 ms If 40ms is when it's extremely good, then I'm glad I don't live where you are! (However you're right about leased lines, E1 where I am is typically 2ms-3ms, but people here are moving to Ethernet based carriers now) Gordon -- _ -- 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] Moving from DSL to T1
On Sun, 2010-09-12 at 15:32 -0700, Kevin Keane wrote: In terms of telephony, a T-1 can make a huge difference over DSL. DSL gives you a lot of raw bandwidth, true, but for voice that really doesn’t matter all that much. Voice calls only take a relatively small amount of bandwidth anyway; you can fit dozens of concurrent calls into a DSL or T-1. When used strictly for telephony (non-VoIP), a T-1 is designed for 24 concurrent calls, each one takes up 56kbit. For VoIP use, most providers tell you that a phone call takes up about 80kbit/s. What really matters is the latency, and T-1 is a huge improvement over DSL in that area. The easiest way to measure latency is the ping time to a server that is “close to you” Internet-wise. A DSL has latencies of between 40ms (if it’s extremely good and not too many other people are using it) and 1000ms (if there is a problem somewhere). A good T-1 may have latencies as low as 5 ms or so. Also, with a T-1 the bandwidth is guaranteed and bidirectional. With a DSL line, you may get burstable bandwidth – you don’t actually have that bandwidth, you just get to compete for excess bandwidth with your neighbors. You consider 40ms extremely good??? Either your isp or youself must have a considerable number of hops to cross. At home (cheap abo) i got following transit delays (round trip) national 15 ms international 17-35 transatlatic or satelite is above 200ms At work national 3-4 ms international 20-25 ms Latency also is the reason VoIP does not work at all over satellite connections even though they tend to have plenty of bandwidth. Please define does not work at all over satellite ??? Sure, it is not studio HIFI quality, but is th same quality as you get from official commercial telco providers. We still have voip over S-band and X-band satelites running NOW between NL and afghanistan. All the people are more than satisfied. To answer the OP’s question: assuming that you will be using the T-1 for mixed VoIP and data (the most likely scenario in this case), a T-1 is really not much different from a DSL line. Both provide you with IP connectivity. Just make sure that QoS is set up correctly on your router and firewall to give priority to VoIP calls. If you are using VoIP and DSL concurrently and your router/firewall supports that configuration, you may also need to modify routing tables to make sure calls go in and out over the correct link. Other big advantage of T1 above (commercial end-user grade) DSL lines, is that you have a higher upload bandwith Theyoften sell it as 1Mb up, but most of the time you'll have to be satisfied with the half. But with strong compression that is still more than enough. -- _ -- 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] Moving from DSL to T1
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Sunday, September 12, 2010 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving from DSL to T1 On Sun, 2010-09-12 at 15:32 -0700, Kevin Keane wrote: In terms of telephony, a T-1 can make a huge difference over DSL. DSL gives you a lot of raw bandwidth, true, but for voice that really doesn't matter all that much. Voice calls only take a relatively small amount of bandwidth anyway; you can fit dozens of concurrent calls into a DSL or T-1. When used strictly for telephony (non-VoIP), a T-1 is designed for 24 concurrent calls, each one takes up 56kbit. For VoIP use, most providers tell you that a phone call takes up about 80kbit/s. What really matters is the latency, and T-1 is a huge improvement over DSL in that area. The easiest way to measure latency is the ping time to a server that is close to you Internet-wise. A DSL has latencies of between 40ms (if it's extremely good and not too many other people are using it) and 1000ms (if there is a problem somewhere). A good T-1 may have latencies as low as 5 ms or so. Also, with a T-1 the bandwidth is guaranteed and bidirectional. With a DSL line, you may get burstable bandwidth - you don't actually have that bandwidth, you just get to compete for excess bandwidth with your neighbors. You consider 40ms extremely good??? Either your isp or youself must have a considerable number of hops to cross. At home (cheap abo) i got following transit delays (round trip) national 15 ms international 17-35 transatlatic or satelite is above 200ms At work national 3-4 ms international 20-25 ms (sorry about the weird quoting - Outlook insists on top-posting!) Wow. I think I have to move to the Netherlands. The European telecom landscape is quite a bit different from US, so I'm not completely surprised. The USA no longer has the fastest Internet in the world anyway. My numbers are from an ATT DSL line in California, suburban San Diego county, and just around the corner from the central office. So it is not the distance (with DSL, the distance does make quite a difference). On the other hand, there are several hops just to get to the Internet backbone. Your work numbers sound like what I have seen with a T-1 here. Latency also is the reason VoIP does not work at all over satellite connections even though they tend to have plenty of bandwidth. Please define does not work at all over satellite ??? Sure, it is not studio HIFI quality, but is th same quality as you get from official commercial telco providers. We still have voip over S-band and X-band satelites running NOW between NL and afghanistan. All the people are more than satisfied. ** Should have been more specific. I was talking about Internet over satellite in the USA. I believe those are geostationary TV satellites. I am not familiar with S-band and X-band, but assume they are in lower orbit. That would explain how it can work for you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Correct queue agi syntax in 1.6.2.11
Hello list,
what is the correct syntax ?
exten = s,n,Queue(${queuename}${timeout},cleanpickup.agi^${CHANNEL})
[Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to
execute
'/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-017a':
File does not exist.
Kind regards,
Jonas.
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[asterisk-users] How to send SMS to Gigaset phones ?
Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as This phone system will be stopped in 5mn for maintenance to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). I could successfully send 1 short text message to 1 SIP hardphones (using sipsak) (I've not tried to send the same message to several phones and to see I could erase old messages form phone's screen) but failed with Gigasets. Which tool shall I use ? smsq ? sipsak ? It this matters, I'm using 1.4.35. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.
On 11 September 2010 20:36, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
On 09/09/10 17:59, Steve Davies wrote:
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording filename. But, the macro runs on the calling channel, and I
need the called channel - Is this accessible?
Thanks,
Steve
Where ever the MixMonitor recording is done add in the ${CHANNEL}
variable to the filename parameter. Or even add in the line below to the
context that contains Dial(QueueName).
For example:
exten =
s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49)
[snip]
Why is it you require the answering channel in the recording filename?
There may be an easier way to get you what you need. And a quick copy
pasta of the dial plan you're using could be handy.
I will look at putting together an example of the dialplan.
We need the answering channel so the we can identify which handset was
recorded (which agent took the call). At present we have caller-id and
dialled extension, but an extension might be ringing any one of 10
phones, and the recording needs to be associated with that
handset/agent.
Regards,
Steve
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[asterisk-users] High volume BLF - Suggestions?
Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- _ -- 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] Which voicemail file format is the most widely understood ?
Hi, In voicemail.conf you can choose among several file format (wav, wav49 and gsm) with which voicemail are saved. Which one the is the most widely read by Windows, Mac and Linux PC media players ? Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.
On 09/13/2010 10:11 AM, Steve Davies wrote:
On 11 September 2010 20:36, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
On 09/09/10 17:59, Steve Davies wrote:
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.comwrote:
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording filename. But, the macro runs on the calling channel, and I
need the called channel - Is this accessible?
Thanks,
Steve
Where ever the MixMonitor recording is done add in the ${CHANNEL}
variable to the filename parameter. Or even add in the line below to the
context that contains Dial(QueueName).
For example:
exten =
s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49)
[snip]
Why is it you require the answering channel in the recording filename?
There may be an easier way to get you what you need. And a quick copy
pasta of the dial plan you're using could be handy.
I will look at putting together an example of the dialplan.
We need the answering channel so the we can identify which handset was
recorded (which agent took the call). At present we have caller-id and
dialled extension, but an extension might be ringing any one of 10
phones, and the recording needs to be associated with that
handset/agent.
Regards,
Steve
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by agent that took the call) here but since our asterisk call
recording is a separate server to the ones dealing with queues I'll be
looking at tie-ing the agent to the call recording via a unique call ID
in a database rather than in the filename itself. I'll post my
findings/solution to the list if you would like?
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Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.
On 13 September 2010 11:07, Antonio Berrios anto...@sheffieldcitytaxis.com wrote: Gotcha. Yeah, I'm looking at implementing that (searching call recordings by agent that took the call) here but since our asterisk call recording is a separate server to the ones dealing with queues I'll be looking at tie-ing the agent to the call recording via a unique call ID in a database rather than in the filename itself. I'll post my findings/solution to the list if you would like? Sounds like a plan :) I may try to modify Asterisk to send more parameters to the MONITOR_EXEC script if requested to do so. UniqueID, CDR(channel) and CDR(dstchannel) seem like obvious choices. I'll post results here if I manage it. Regards, Steve -- _ -- 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] Upgrade from 1.4 to 1.6 : problems with realtimemysql
This is a problem with extconfig.conf - not your res_ or cdr_ ones. In your case - extconfig.conf probably contained something like 'sippeers = mysql,MyDBase,sippeers'. The 'problem' is that the middle parameter is no longer for the database name - it is for the context in res_mysql.conf. So, the above now becomes 'sippeers = mysql,general,sippeers'. Give that a go... Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 08 September 2010 15:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql Hello, in asterisk 1.4.30 all realtime configurations go well. In asterisk 1.6.2.11 the following appears on CLI : [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf) [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf) res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = MyDBase dbuser = asterisk dbpass = mysecret dbport = 3306 dbsock = /tmp/mysql.sock requirements=warn ; or createclose or createchar What do I need to change to be conform asterisk 1.6 ?! Reloading, restarting asterisk and restarting my CentOS-server all doesn't help. Jonas. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
We have good success with Grandstream, but even though we monitor 100 phones, I don't think we get the amount of NOTIFY messages you describe. But you should try them. Ron Op 13-09-10 11:56, Steve Davies schreef: Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- NeoNova BV innovatieve internetoplossingen http://www.neonova.nl Galjoen 6 1113 GS Diemen KvK Amsterdam 34151241 Op dit bericht is de volgende disclaimer van toepassing: http://www.neonova.nl/maildisclaimer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing voicemail.conf file format list
Hi, In voicemail.conf.sample, you can read this: format=wav49|gsm|wav ; WARNING: ; If you change the list of formats that you record voicemail in ; when you have mailboxes that contain messages, you _MUST_ absolutely ; manually go through those mailboxes and convert/delete/add the ; the message files so that they appear to have been stored using ; your new format list. If you don't do this, very unpleasant ; things may happen to your users while they are retrieving and ; manipulating their voicemail. ; ; In other words: don't change the format list on a production system ; unless you are _VERY_ sure that you know what you are doing and are ; prepared for the consequences. What does manually go through those mailboxes and convert/delete/add the ; the message files so that they appear to have been stored using ; your new format list exactly imply here ? As this statement is written, it seems there is a trap too complex to detail which surprises me a bit. Comments ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
2010/9/13 Steve Davies davies...@gmail.com Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- _ -- 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] Which voicemail file format is the most widely understood ?
Hi, On 09/13/2010 10:57 AM, Olivier wrote: Hi, In voicemail.conf you can choose among several file format (wav, wav49 and gsm) with which voicemail are saved. Which one the is the most widely read by Windows, Mac and Linux PC media players ? I use wav49. It is compatible with Windows and Linux at least. Don't know about Mac though. I used to use gsm - but only some apps play it on Linux - and I could only find QuickTime Player to play it in Windows. Simple 'wav' appears to be incompatible (at least not directly) with Windows. Sebastian Suggestions ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
On 13 September 2010 11:43, Olivier oza_4...@yahoo.fr wrote: 2010/9/13 Steve Davies davies...@gmail.com [snip] Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. [snip] The SUBSCRIBE is handled fine, it is the NOTIFY messages that cause a problem. Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing voicemail.conf file format list
Hi, On 09/13/2010 11:34 AM, Olivier wrote: Hi, In voicemail.conf.sample, you can read this: format=wav49|gsm|wav ; WARNING: ; If you change the list of formats that you record voicemail in ; when you have mailboxes that contain messages, you _MUST_ absolutely ; manually go through those mailboxes and convert/delete/add the ; the message files so that they appear to have been stored using ; your new format list. If you don't do this, very unpleasant ; things may happen to your users while they are retrieving and ; manipulating their voicemail. ; ; In other words: don't change the format list on a production system ; unless you are _VERY_ sure that you know what you are doing and are ; prepared for the consequences. What does manually go through those mailboxes and convert/delete/add the ; the message files so that they appear to have been stored using ; your new format list exactly imply here ? It sounds like you have to shutdown Asterisk, find the old voicemail messages stored on the server in their respective directories, and convert all of them to the new file/audio format, then modify voicemail.conf and then re-start Asterisk. At least that's what it sounds like to me. In other words, it seems that Asterisk will get confused if it finds old voicemail messages in the storage with a format different from what voicemail.conf tells it to expect. I suppose if you go and delete all the old voicemail messages before changing the format in voicemail.conf, it will work equally well. However, your users might not be very pleased :-) Sebastian As this statement is written, it seems there is a trap too complex to detail which surprises me a bit. Comments ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
Hello, Am 13.09.10 11:56, schrieb Steve Davies: Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? we only use the spa962 and spa525 for this, but didnt have any problems so far. 2) Is there a different (standard) way to send BLF and allow directed pickups? have a look at www.fop2.com maybe this could be interesting for you? its a web based solution to see the state of extensions and do a pickup or transfer calls. best regards steve -- _ -- 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] High volume BLF - Suggestions?
On 13 September 2010 12:16, Stefan Schmidt s...@sil.at wrote: Hello, Am 13.09.10 11:56, schrieb Steve Davies: Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? we only use the spa962 and spa525 for this, but didnt have any problems so far. We find that the spa962 works pretty well with one sidecar (32 buttons) we only have one spa932 sidecar to hand though. The spa525 seems to not like our sidecar configuration, and allows only 59 subscriptions, so the last few lights do not work, and after a while it decides to reboot regularly. Not quite sure how to debug that one :( The spa504 works pretty well with a sidecar, but crashed under load over a weekend. I have updated 7.4.3 to 7.4.4 firmware and we're trying again. 2) Is there a different (standard) way to send BLF and allow directed pickups? have a look at www.fop2.com maybe this could be interesting for you? its a web based solution to see the state of extensions and do a pickup or transfer calls. Yes, we have a web-based BLF solution just like FOP, but some end-users have very specific requirements *sigh* :) Thanks for the input. Cheers, Steve -- _ -- 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] High volume BLF - Suggestions?
As a side note to this - do NOT try and use Aastra's - as they tend to crash after 50 BLF's! Also, could you please send me (perhaps off-list to a...@datavox.co.uk) your Yealink T28 findings - as I am a beta tester for them? Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 13 September 2010 11:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High volume BLF - Suggestions? 2010/9/13 Steve Davies davies...@gmail.com Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
On 13/09/10 11:03 PM, Steve Davies wrote: On 13 September 2010 11:43, Olivieroza_4...@yahoo.fr wrote: 2010/9/13 Steve Daviesdavies...@gmail.com [snip] Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. [snip] The SUBSCRIBE is handled fine, it is the NOTIFY messages that cause a problem. Yeah, I would have said Snoms too, not because I know them to work in that situation, but because they seem to be sorting various people's problems via firmware upgrades pretty regularly. I'd advise flicking the people over at Snom and email, then posting back here with your results :) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to send SMS to Gigaset phones ?
On Mon, 13 Sep 2010, Olivier wrote: Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as This phone system will be stopped in 5mn for maintenance to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). I could successfully send 1 short text message to 1 SIP hardphones (using sipsak) (I've not tried to send the same message to several phones and to see I could erase old messages form phone's screen) but failed with Gigasets. Which tool shall I use ? smsq ? sipsak ? It this matters, I'm using 1.4.35. I think you're stuffed unless you use the analogue side of the Siemens gigaset base stations - I'm fairly sure they can only take SMS messages that way. I'd suggest that you really ought to be sending SIP messages using the SendText() command, however don't seem to respond to that anyway, and other phones seem to have different ideas about what to do with them too (to the extent that I abandoned it after some experiments - good idea if you can stick to one phone type, bad if you've got many) Gordon -- _ -- 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] Moving from DSL to T1
On Mon, Sep 13, 2010 at 1:32 AM, Kevin Keane subscript...@kkeane.comwrote: My numbers are from an ATT DSL line in California, suburban San Diego county, and just around the corner from the central office. So it is not the distance (with DSL, the distance does make quite a difference). On the other hand, there are several hops just to get to the Internet backbone. Lots of misconceptions in this thread. I'll limit this discussion to what I know - ADSL, T1, and cable as delivered by US telcos/ISPs. I just measured my Denver, CO Qwest DSL line. I have a short DSL line - about 4 city blocks, but then, like most metro areas served by Qwest, once my connection ends up in the exchange, I'm backhauled about 30 miles via ATM to the actual edge router(s) that serve the metro area (the edge router is not in the central office).: 10 packets transmitted, 10 packets received, 0.0% packet loss round-trip min/avg/max/stddev = 28.646/31.359/33.828/1.603 ms This is pretty typical of Qwest DSL in Colorado and Wyoming. I've literally run hundreds of DSL sites in these states, and find that the DSL product is a great bargain for business (I usually didn't run it to Qwest's edge router but to my own corporate routers, as basically an ATM circuit that just happens to be delivered on DSL). I ran a large video conferencing network on it without any issues (H323). The reason for the times being around 30 ms round trip is that Qwest uses interleaving on the DSL circuit. It's on for a reason (it makes TCP streams faster - instead of retransmitting whole 1500 byte TCP packets, the system retransmits the 53 byte ATM cell instead. It might have to do that several times, for several packets that get corrupted, but it will still be faster than TCP dealing with the loss. Of course VoIP doesn't use TCP, nor does it need 100% guaranteed packet delivery. But this also can help with jitter, depending on what else is on the line. Typical T1s do not do this interleaving (they are better engineered, regardless of whether they are delivered on true T1 media or backhauled via HDSL - either way, the packets make it out the other end a lot more reliably than they do on ADSL connections). For DSL, there are no shared connections between the telephone exchange and the home - everything there is dedicated. Cable shares a connection with your neighbors. T1 is dedicated to the Exchange as well (burstable or dedicated bandwidth both). At the Exchange, DSL is aggregated on a large (OC3 typically) ATM circuit to get transported to the ISP (or telco's own) edge router. This is shared bandwidth with everyone else on the same DSLAM. A point-to-point T1 has dedicated bandwidth (no aggregation) to the ISP. A frame relay, MPLS, or ATM T1 has the amount of dedicated bandwidth you pay for (in my experience, usually none, as that's cheapest) with the rest of the bandwidth provided if there is capacity on that ATM circuit between the central office and the ISP/telco edge router. Cable is also usually aggregated at some intermediate point, similar to DSL. Once connected to the ISP edge router, all circuits are aggregated out to the internet backbone via a connection smaller than the sum of all the subscriber bandwidths. So, frame relay, MPLS, and ATM T1s - as typically ordered - often function like DSL, with the same choke points. Even a point-to-point T1 that goes to the internet however will hit a choke point and might have packet loss - no matter how much you paid for the T1 (the internet backbone itself has packet loss and choke points). Cable has an additional choke point (subscriber loop). Now, hopefully the ISP and telco have engineered everything to not have significant choke points and you'll never have a capacity problem (the same goes with their peering connections - hopefully they, too, are big enough). But even an MPLS burstable T1 could perform badly at high capacity times. Most of these choke points (such as the DSL DSLAM to edge router) do know to not let one user monopolize the uplink, but to let each user have the same approximate capacity when there is congestion. So someone with say only one VoIP call won't experience packet loss or changes, while another user downloading a movie will see a slower download. (in the IP world, this is called fair queuing - in the ATM world, it goes by other names; they idea is that you should give everyone the same amount of possible bandwidth in a congestion situation, even if they aren't using all of that bandwidth). Most of these technologies do not let you apply QoS to the choke points effectively. So you are left with point-to-point T1s or other T1s that you buy with guaranteed (more expensive) bandwidth. But you can't QoS across the internet. Sure, you can do some traffic shaping on a DSL line, and it'll work good most of the time, but there are no guarantees with guaranteed bandwidth! So, the only way to gaurantee packet delivery is to build your voice IP network like you
Re: [asterisk-users] How to send SMS to Gigaset phones ?
On Mon, Sep 13, 2010 at 3:29 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Mon, 13 Sep 2010, Olivier wrote: how to send SMS to Gigaset phones ? I dimly recall someone doing this and publishing a page of script... but where? Look using Google back from 2004-2008 something like sms gigaset script and I'll bet you will find something. I know someone did this (I was going to use it but changed my mind). r -- _ -- 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] How to send SMS to Gigaset phones ?
2010/9/13 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 13 Sep 2010, Olivier wrote: Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as This phone system will be stopped in 5mn for maintenance to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). I could successfully send 1 short text message to 1 SIP hardphones (using sipsak) (I've not tried to send the same message to several phones and to see I could erase old messages form phone's screen) but failed with Gigasets. Which tool shall I use ? smsq ? sipsak ? It this matters, I'm using 1.4.35. I think you're stuffed unless you use the analogue side of the Siemens gigaset base stations - I'm fairly sure they can only take SMS messages that way. I'd suggest that you really ought to be sending SIP messages using the SendText() command, however don't seem to respond to that anyway, and other phones seem to have different ideas about what to do with them too (to the extent that I abandoned it after some experiments - good idea if you can stick to one phone type, bad if you've got many) Form changelogs, you can see: Firmware Update 04/2009 Version V02184: --- New features: - E-mail viewer (with C47H, S45, S67H, S68H handsets) - Mute function. Turn off the handset's microphone during an external call with the left display key. - Send and receive SMS messages via VoIP* snip * provider-dependent The bad thing is the way to use this new SMS-VOIP handling capability is not much documented. Gordon -- _ -- 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] Problem with new AEX800 card dying because of interrupt problems
Christian Weeks c...@weeksfamily.ca writes: Hello I purchased an AEX800 card to replace the ageing cheap channel bank/T1 card solution a few months ago, assuming that it would be a more robust solution for my small scale phone system. However, it appears to be anything but that. Originally implemented as a XEN dom-u virtual machine on a large server class machine, using PCI passthrough to pass the AEX800 and a small older TDM400, then recently migrated to the dom-0, the aex800 has continued to experience interrupt errors: wctdm24xxp :04:08.0: Missed interrupt. Increasing latency to 8 ms in order to compensate. wctdm24xxp :04:08.0: ERROR: Unable to service card within 25 ms and unable to further increase latency. Can you do a at /proc/interrupts? /Benny -- _ -- 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] Force ip disconnect after register?
Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box with reg attempts as long as they maintain their initial connection or I reset the state tables on the firewall. This is very bad. Is there a way to force the connection to drop and reconnect after let's say 50 attempts. Thanks for any input. Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing voicemail.conf file format list
On Monday 13 September 2010 06:07:09 Sebastian wrote: On 09/13/2010 11:34 AM, Olivier wrote: In voicemail.conf.sample, you can read this: format=wav49|gsm|wav ; WARNING: ; If you change the list of formats that you record voicemail in ; when you have mailboxes that contain messages, you _MUST_ absolutely ; manually go through those mailboxes and convert/delete/add the ; the message files so that they appear to have been stored using ; your new format list. If you don't do this, very unpleasant ; things may happen to your users while they are retrieving and ; manipulating their voicemail. ; ; In other words: don't change the format list on a production system ; unless you are _VERY_ sure that you know what you are doing and are ; prepared for the consequences. What does manually go through those mailboxes and convert/delete/add the ; the message files so that they appear to have been stored using ; your new format list exactly imply here ? It sounds like you have to shutdown Asterisk, find the old voicemail messages stored on the server in their respective directories, and convert all of them to the new file/audio format, then modify voicemail.conf and then re-start Asterisk. At least that's what it sounds like to me. In other words, it seems that Asterisk will get confused if it finds old voicemail messages in the storage with a format different from what voicemail.conf tells it to expect. You must also remove all message content with a format that is NOT in your new list (i.e. if you remove a format from the list, you must also remove all recordings in the voicemail hierarchy with that format). And really, this is the critical step for most messages. I believe only forwarding messages with prepend would be a problem for not creating new files. I suppose if you go and delete all the old voicemail messages before changing the format in voicemail.conf, it will work equally well. However, your users might not be very pleased :-) That will also work. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
Steve Grandstream has a new services GXP-21XX coming out they may work for your. We have been a beta tester and the BLF on these seem to work much better then the GXP-20XX units. I do not have the side cars in stock right now so I don't know how they work with it but you can put at least two for about 112 addtional blf keys. Bryant From: Olivier oza_4...@yahoo.fr Sent: Monday, September 13, 2010 6:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] High volume BLF - Suggestions? 2010/9/13 Steve Davies davies...@gmail.com Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- -- _ -- 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] Force ip disconnect after register?
On Mon, Sep 13, 2010 at 11:22 AM, Bryant Zimmerman brya...@zktech.com wrote: Is there a way to drop a ip connection to asterisk after a number of register attempts. Not within Asterisk. Google fail2ban -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force ip disconnect after register?
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote: Is there a way to drop a ip connection to asterisk after a number of register attempts. Consider writing a filter for fail2ban [http://www.fail2ban.org/] that works on the Asterisk logs? -- _ -- 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] SIP softphones answer but do not connect...
On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote: On 11/09/10 12:44 PM, Carlos Chavez wrote: The past few days I started having a problem with a small call center setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam is configured to auto answer the call. The problem is that the agents claim that they get a call but no audio. From the logs I can see that it is calling the agent phone but after 10 seconds (the queue timeout for pickup) I get the message that nobody answered and the call is sent to the next available agent. This can happen with up to three agents (the third finally answers the call). This has happened at least 20 times in the past two days. At first the supervisor thought that the same call was ringing on three different agents at once but the logs say that the first two do not answer and the third does. What strategy are you using for the Queue? We are using Least Recent at the moment. Why would queue strategy impact this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] PostgreSQL is asterisk friendly with it?
As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Is there any down side to PostgreSQL over mySQL or will it be a big win? Our database servers are linux but we access them from asterisk as well as windows are there any thing to be concerned with there? I use c#, vb.net and mono to do a lot of our stuff are there any issues I should know about? Thanks Bryant -- _ -- 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] Correct queue agi syntax in 1.6.2.11
Hello,
can anyone please tell me how I can give arguments to my AGI script ?!
I think asterisk sees the name of the AGI + the channel as one filename,
and of course this file then does not exist.
Jonas.
On 09/13/2010 10:26 AM, Jonas Kellens wrote:
Hello list,
what is the correct syntax ?
exten = s,n,Queue(${queuename}${timeout},cleanpickup.agi^${CHANNEL})
[Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed
to execute
'/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-017a':
File does not exist.
Kind regards,
Jonas.
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Re: [asterisk-users] SIP softphones answer but do not connect...
What strategy are you using for the Queue? We are using Least Recent at the moment. Why would queue strategy impact this? Carlos: I had similar issues, caused by a setting somewhere in the advanced section of eyeBeam. Something about Disconnect if no audio received for x seconds. We turned this option off, and things went back to normal. Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.
On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote:
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by agent that took the call) here but since our asterisk call
recording is a separate server to the ones dealing with queues I'll be
looking at tie-ing the agent to the call recording via a unique call ID
in a database rather than in the filename itself. I'll post my
findings/solution to the list if you would like?
Sounds like a plan :)
I may try to modify Asterisk to send more parameters to the
MONITOR_EXEC script if requested to do so. UniqueID, CDR(channel) and
CDR(dstchannel) seem like obvious choices.
I'll post results here if I manage it.
What we do is get the ${MEMBERINTERFACE} variable from the Queue and
then rename the file in the h extension. To get this variable you need
to set setinterfacevar=yes in your queue definition. We then run an
AGI from the h extension to rename and move the file.
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Director de Tecnología
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Re: [asterisk-users] PostgreSQL is asterisk friendly with it?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Monday, September 13, 2010 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PostgreSQL is asterisk friendly with it? As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Is there any down side to PostgreSQL over mySQL or will it be a big win? Our database servers are linux but we access them from asterisk as well as windows are there any thing to be concerned with there? I use c#, vb.net and mono to do a lot of our stuff are there any issues I should know about? Thanks Bryant In my experience, Asterisk plays much more nicely with MYSQL than POSTGRESQL. PostgreSQL is supported as ODBC vs NATIVE support for MYSQL. The downside that I'm aware of is that certain features of MYSQL aren't directly portable to PostgreSQL (blobs for example). Linux could care less. IMO, Asterisk will be a larger hurdle than C#, vb.net or mono to jump over. This is all in 1.4/1.6; some changes may have modified these answers regarding 1.8. -- _ -- 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] SIP softphones answer but do not connect...
can you state your internet connection your agents are on?and one more thing.. how are the members positioned into the Queue? static? Dynamic? single station and call forwarding (find me follow me extension in the queue)? do you get call waiting override with Auto Answer? -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: cur...@telecomabmex.com To: asterisk-users@lists.digium.com Date: Mon, 13 Sep 2010 10:44:35 -0500 Subject: Re: [asterisk-users] SIP softphones answer but do not connect... On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote: On 11/09/10 12:44 PM, Carlos Chavez wrote: The past few days I started having a problem with a small call center setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam is configured to auto answer the call. The problem is that the agents claim that they get a call but no audio. From the logs I can see that it is calling the agent phone but after 10 seconds (the queue timeout for pickup) I get the message that nobody answered and the call is sent to the next available agent. This can happen with up to three agents (the third finally answers the call). This has happened at least 20 times in the past two days. At first the supervisor thought that the same call was ringing on three different agents at once but the logs say that the first two do not answer and the third does. What strategy are you using for the Queue? We are using Least Recent at the moment. Why would queue strategy impact this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct queue agi syntax in 1.6.2.11
On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote: Hello, can anyone please tell me how I can give arguments to my AGI script ?! I think asterisk sees the name of the AGI + the channel as one filename, and of course this file then does not exist. In Asterisk 1.4 you use the | (pipe) and in 1.6 you use a , (coma). So: runme.agi|parameter or runme.agi,parameter -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] PostgreSQL is asterisk friendly with it?
On Mon, 13 Sep 2010, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Monday, September 13, 2010 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PostgreSQL is asterisk friendly with it? As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Is there any down side to PostgreSQL over mySQL or will it be a big win? Our database servers are linux but we access them from asterisk as well as windows are there any thing to be concerned with there? I use c#, vb.net and mono to do a lot of our stuff are there any issues I should know about? Thanks Bryant In my experience, Asterisk plays much more nicely with MYSQL than POSTGRESQL. PostgreSQL is supported as ODBC vs NATIVE support for MYSQL. The downside that I'm aware of is that certain features of MYSQL aren't directly portable to PostgreSQL (blobs for example). Linux could care less. IMO, Asterisk will be a larger hurdle than C#, vb.net or mono to jump over. This is all in 1.4/1.6; some changes may have modified these answers regarding 1.8. Say what? I just looked at the 1.4 sources and PostgreSQL is NOT supported as ODBC. Those are direct library calls into libpq. Bryant, I've been using PostgreSQL with asterisk since I started using asterisk a number of years ago. There have been exactly ZERO issues with it. Vince. -- Michigan VHF Corp. http://www.nobucks.net/ http://www.CDupe.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.
On 13 September 2010 16:58, Carlos Chavez cur...@telecomabmex.com wrote:
On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote:
On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Gotcha. Yeah, I'm looking at implementing that (searching call
recordings by agent that took the call) here but since our asterisk call
recording is a separate server to the ones dealing with queues I'll be
looking at tie-ing the agent to the call recording via a unique call ID
in a database rather than in the filename itself. I'll post my
findings/solution to the list if you would like?
Sounds like a plan :)
I may try to modify Asterisk to send more parameters to the
MONITOR_EXEC script if requested to do so. UniqueID, CDR(channel) and
CDR(dstchannel) seem like obvious choices.
I'll post results here if I manage it.
What we do is get the ${MEMBERINTERFACE} variable from the Queue and
then rename the file in the h extension. To get this variable you need
to set setinterfacevar=yes in your queue definition. We then run an
AGI from the h extension to rename and move the file.
Excellent. thank you.
:)
Steve
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Re: [asterisk-users] Force ip disconnect after register?
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote: Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box with reg attempts as long as they maintain their initial connection or I reset the state tables on the firewall. This is very bad. Is there a way to force the connection to drop and reconnect after let's say 50 attempts. Not an exact answer to your question, but if the attacker is using svwar (part of SIPVicious), setting alwaysauthreject=yes in sip.conf will make the probing stop after only TWO tries. svwar first tries registering a few longish, random extensions before it begins a sequential or dictionary scan, to see how you handle unknown extensions. With alwayauthreject set, svwar just gives up, complaining: ERROR:TakeASip:SIP server replied with an authentication request for an unknown extension. Set --force to force a scan. I still see 3-4 attempts per week from various sites, but now they stop after just two failed registration attempts. Saves lots of wear and tear on my DSL. I still run fail2ban, but after setting alwaysauthreject a few months ago nothing has passed its threshold. And nothing seems to have broken, either. -- Barry -- _ -- 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] PostgreSQL is asterisk friendly with it?
On Mon, Sep 13, 2010 at 12:31:55PM -0400, Vince Vielhaber wrote: change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully [snippage and probably off topic] Why are you worried re the future of MySQL and it's new owners Oracle. a) MySQL is open source so anyone can take a fork and continue development. b) Oracle own InnoDB already which is the main storage engine for MySQL. c) Oracle dont have any low end DB products for start-ups etc. Developer licenses may be free, but commercial use certainly isn't. d) MySQL is now a business division within Oracle and MySQL (.com) makes money. e) Google have contributed a lot of code for MySQL v6, I'm sure they'd take it on if Oracle in madness decided to drop it. f) I'm sure Oracle will push Oracle on Sun hardware/Solaris for high-end DB platforms and optimise the two so the performance stats look great and so they'll develop a migration platform so high-end MySQL users can easily migrate to Oracle when the need arises. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- 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] doing dnsmgr_lookup
At 11:38 PM 9/12/2010, you wrote: my CLI is spammed with : [Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld' How can I turn this off ?! I just change the 4 in the source code to 14 so it doesn't show up till you add a lot of Vs. It only occurs in one place in the code so it's easy to find. Ira -- _ -- 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] PostgreSQL is asterisk friendly with it?
I agree with Steve, I've had a couple of people speak with me about concerns of MySQL's new ownership, but I stand by the same reasons Steve doesMySQL will not be going anywhere, and if it does, someone'll just fork and keep it going under a new name. On Mon, Sep 13, 2010 at 11:46 AM, Steve Kennedy steve-aster...@gbnet.net wrote: On Mon, Sep 13, 2010 at 12:31:55PM -0400, Vince Vielhaber wrote: change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully [snippage and probably off topic] Why are you worried re the future of MySQL and it's new owners Oracle. a) MySQL is open source so anyone can take a fork and continue development. b) Oracle own InnoDB already which is the main storage engine for MySQL. c) Oracle dont have any low end DB products for start-ups etc. Developer licenses may be free, but commercial use certainly isn't. d) MySQL is now a business division within Oracle and MySQL (.com) makes money. e) Google have contributed a lot of code for MySQL v6, I'm sure they'd take it on if Oracle in madness decided to drop it. f) I'm sure Oracle will push Oracle on Sun hardware/Solaris for high-end DB platforms and optimise the two so the performance stats look great and so they'll develop a migration platform so high-end MySQL users can easily migrate to Oracle when the need arises. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- 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] How to send SMS to Gigaset phones ?
Hi Olivier, I remember having had a similar discussion a few years ago. I will paste my postings from around May 2007 further down. First, I did not try sending SMS over VOIP to the phone, just over Voip to an ATA and then over analogue line (or ISDN) to the phone. So I have no idea wether the new Gigaset VoIP phones will to 1200 baud mumbo SMS phone service over a Sip voice channel or if Gigaset invented something better by now. You will have to try yourself. As for Gigaset phones connected via (at least one cable of ;- ) landline, you can send SMS messages to those with smsq. In theory that should also work on other landline SMS capable phones. Am Montag, den 13.09.2010, 11:04 +0200 schrieb Olivier: Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as This phone system will be stopped in 5mn for maintenance to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm not looking for way to send SMS from handsets). == Message 1 (from myself, 2007-May-22) The SMSq stuff is for landline-type SMS, like those that never became really popular here in Europe ;-) I do not know of any SIP hardphone that supports them, but regular analog and ISDN handsets behind a SIP-to-analog/ISDN gateway work for me. The point of this SMS transfer method is calling the destination handset with a certain callerid set (which differs between countries - whatever number the telco prefers to choose - this can also be configured in the phone). The phone will not ring but instead immediately answer the call and receive the short message at 1200bps whatever modem standard they chose to use. For sending SMS, the handset will call a similarly telco-provided number (premium-rate numbers here in Germany - maybe that is the reason for the lack of popularity of this service) and do that 1200bps talk. If you still think you can make use of it, make sure to call smsq with the user id that asterisk is running as. That _might_ already do the trick. If you do not get it running, ask again - I might have a working setup somewhere around ;-) Nevertheless, for me, landline SMS is a PITA. The only great thing is you can upload Ringtones to Siemens gigaset phones. == Message 2 (from myself, 2007-May-22) Just to get you started, try this: Find out which user asterisk runs as. Get a shell for that user. Run (all in one line) smsq --mt --oa=321 --mttx-callerid=01930101 --mttx-channel=SIP/abcde message text goes here where 321 will displayed as sender id on the handset, and 01930101 will have to replaced by the mobile center known to your phone, plus 1 at the end - the German T-Com seems to use 0193010, and this setting works for me. Further, SIP/abcde must be the channel that a SMS-capable handset is available on: If you have some ATA with a DECT handset connected, or similar, use the channel name exactly as you would in the Dial() command. First thing to find out is if this works. Be sure to have asterisk in extra-verbose running a console to see what happens. If the mobile handset rings (instead of getting the SMS) either the 01930101 number has not been set correctly or it probably is not compatible with Asterisk SMS. Once you get this far, you would need the other way round. When your mobile phone tries to _send_ a text message, it will go to 01930100 (sms center number plus 0). You will have to care for that in your extensions.conf, like this exten = 01930100,1,Wait(2) exten = 01930100,2,Answer() exten = 01930100,3,Wait(2) exten = 01930100,4,SMS(01930100,as) exten = 01930100,5,Wait(2) exten = 01930100,6,Hangup() In my experience those Wait(2) improve reliability over internet connections, they probably are superfluous if you have reliable low-latency LAN. For me, they made the difference between 10/100 and 95/100 successfuly sent messages. You will have to write your own scriptwork to play with the files that will be created from those commands. Their structure is simple, you will find out. Sending EMS (for ringtones and bitmaps) is a bit more complex, you will need the UDH flag for that. I think I documented that once on this ML but am not sure. However, it is possible with some Siemens Gigaset devices, and pictures or monophonic ringtones. == Message 3 (2006-Nov-12) can be found at http://www.mail-archive.com/asterisk-...@lists.digium.com/msg24205.html with an example of how to send an EMS (message with picture attached). This worked with both monochrome pictures and single-track MIDI ringtones on my Gigaset S1 back then. Never got around to sending multi-track ringtones though. == Best regards Anselm -- _ -- 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
Re: [asterisk-users] SIP softphones answer but do not connect...
On Mon, 2010-09-13 at 15:59 +, Tarek Sawah wrote: can you state your internet connection your agents are on? and one more thing.. how are the members positioned into the Queue? static? Dynamic? single station and call forwarding (find me follow me extension in the queue)? do you get call waiting override with Auto Answer? -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308 From: cur...@telecomabmex.com To: asterisk-users@lists.digium.com Date: Mon, 13 Sep 2010 10:44:35 -0500 Subject: Re: [asterisk-users] SIP softphones answer but do not connect... On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote: On 11/09/10 12:44 PM, Carlos Chavez wrote: The past few days I started having a problem with a small call center setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam is configured to auto answer the call. The problem is that the agents claim that they get a call but no audio. From the logs I can see that it is calling the agent phone but after 10 seconds (the queue timeout for pickup) I get the message that nobody answered and the call is sent to the next available agent. This can happen with up to three agents (the third finally answers the call). This has happened at least 20 times in the past two days. At first the supervisor thought that the same call was ringing on three different agents at once but the logs say that the first two do not answer and the third does. What strategy are you using for the Queue? We are using Least Recent at the moment. Why would queue strategy impact this? All agents are on a local LAN with no Internet access. We use realtime for configuration but agents are defined as Static agents (Agent/XXX). -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] High volume BLF - Suggestions?
Steve I have 64 channels being monitored with an SPA962 with two SPA932 sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very happy with this. Latest firmware is a must. HTH Cassius Smith -- _ -- 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] Correct queue agi syntax in 1.6.2.11
On 09/13/2010 06:01 PM, Carlos Chavez wrote: On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote: Hello, can anyone please tell me how I can give arguments to my AGI script ?! I think asterisk sees the name of the AGI + the channel as one filename, and of course this file then does not exist. In Asterisk 1.4 you use the | (pipe) and in 1.6 you use a , (coma). So: runme.agi|parameter Result : [Sep 13 20:10:27] WARNING[19929]: pbx.c:1344 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (AGI(cleanpickup.agi|SIP/329909007906-017a)) [Sep 13 20:10:27] WARNING[19929]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi|SIP/329909007906-017a': File does not exist. or runme.agi,parameter [Sep 13 20:14:59] WARNING[19965]: app_macro.c:302 _macro_exec: No such context 'macro-SIP/329909007906-017a' for macro 'SIP/329909007906-017a' [Sep 13 20:14:59] -- Launched AGI Script /var/lib/asterisk/agi-bin/cleanpickup.agi [Sep 13 20:14:59] opruimenpickup.agi: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied How can this work for you ?! Do you have an example ?! Or Asterisk says it is not the correct delimiter, or it sees my argument as a macro... Some feedback please ! Jonas. -- _ -- 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 way to check against a list of numbers?
Hose hose+aster...@bluemaggottowel.com writes: The most straightforward way would be to just define explicit patterns. Obviously that works, but doesn't seem scalable in terms of maintenance. I don't think that maintaining the list in the dial plan is all that bad, actually. Dump it in its own context and file... If that isn't convenient enough I'd go for the Asterisk database next. Also on the option list is private e164/enum or an SQL database. /Benny -- _ -- 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] PostgreSQL is asterisk friendly with it?
Bryant Zimmerman brya...@zktech.com writes: As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Yes. The ODBC drivers don't really care which database you access. Is there any down side to PostgreSQL over mySQL or will it be a big win? The only issue we have with Postgres is the dump/reload cycle when upgrading database version. This is being fixed in the latest versions though. Our database servers are linux but we access them from asterisk as well as windows are there any thing to be concerned with there? It works fine from Windows as well. /Benny -- _ -- 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] Correct queue agi syntax in 1.6.2.11
On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote: [Sep 13 20:14:59] -- Launched AGI Script /var/lib/asterisk/agi-bin/cleanpickup.agi [Sep 13 20:14:59] opruimenpickup.agi: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied So check that /var/lib/asterisk/agi-bin/cleanpickup.agi is executable by the user under which asterisk is running. -- _ -- 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] Correct queue agi syntax in 1.6.2.11
No that is not the problem, nor was it the question. I found the solution. Apparently you need to place the AGI-script and its arguments between , but the arguments still need to be separated by a comma. Example : exten = s,n,Queue(queuename,myscript.agi,arg1,arg2) If anyone wants to update the very old wiki, be my guest. Kind regards, Jonas. On 09/13/2010 08:25 PM, Roger Burton West wrote: On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote: [Sep 13 20:14:59] -- Launched AGI Script /var/lib/asterisk/agi-bin/cleanpickup.agi [Sep 13 20:14:59] opruimenpickup.agi: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied So check that /var/lib/asterisk/agi-bin/cleanpickup.agi is executable by the user under which asterisk is running. -- _ -- 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] doing dnsmgr_lookup
Hello, anyone on this list knows how to turn these messages off please ?! I have in sip.conf : srvlookup=no and in dnsmgr.conf : [general] enable=no; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200; refresh managed DNS lookups every n seconds ; default is 300 (5 minutes) But I still have these messages... Jonas. On 09/13/2010 08:38 AM, Jonas Kellens wrote: Hello list, my CLI is spammed with : [Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] doing dnsmgr_lookup for 'ssw4.itsp.tld' How can I turn this off ?! dnsmgr.conf : [general] enable=no; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200; refresh managed DNS lookups every n seconds ; default is 300 (5 minutes) Kind regards, Jonas. -- _ -- 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 way to check against a list of numbers?
i have this scenario where i have a marketing department calling USA numbers
excessively and sometimes the leads contain duplicate numbers OR duplicate
customers with different numbers on the other hand we have some numbers that
are black listed the destination should be checked and caller should be
informed in both cases.
the following dialplan would first check if the number is blackliste (from
local MYSQL DB) .. challenge it then continue to MSSQL DB where existing
customers info is located and challenge the phone number against existing
customers to see if the call should go through or not.
exten = _1N.,1,MYSQL(Connect connid localhost localSQLuser password
blacklistDB)exten = _1N.,n,MYSQL(Query resultid_1 ${connid} SELECT COUNT(*)
FROM tbl_BlackList WHERE PhNumber=${EXTEN})exten = _1N.,n,MYSQL(Fetch fetchid1
${resultid_1} ifpresent)exten = _1N.,n,MYSQL(Disconnect ${connid})exten =
_1N.,n,GotoIF($[${ifpresent} = 0] ?pok:perror);;; IF THE NUMBER EXISTS TELL THE
CALLER THAT IT'S BLACKLISTEDexten = _1N.,n,MYSQL(Clear ${resultid_1})exten =
_1N.,n,MYSQL(Clear ${fetchid1})exten = _1N.,n(perror),Wait(1)exten =
_1N.,n,PlayBack(privacy-blacklisted)exten = _1N.,n,congestion(1)exten =
_1N.,n,HangUpexten = _1N.,n(pok),GoToIf($[${ODBC_CHKAVAIL(${EXTEN})} =
0]?dial:exerror)exten = _1N.,n(dial),GoTo(dial-usa,${EXTEN},1)exten =
_1N.,n(exerror),PlayBack(already-in-db) ;;; PLAY SOUND FILE THE CUSTOMER
ALREADY IN DATABASEexten = _1N.,n,Hangup
you can use the above example to check the number being dialed against your DB
(what ever DBMS you are using) and route it depending on the result of your SQL
query.hope this helps
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
From: benny+use...@amorsen.dk
To: hose+aster...@bluemaggottowel.com
Date: Mon, 13 Sep 2010 20:18:08 +0200
CC: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] A way to check against a list of numbers?
Hose hose+aster...@bluemaggottowel.com writes:
The most straightforward way would be to just define explicit patterns.
Obviously that works, but doesn't seem scalable in terms of maintenance.
I don't think that maintaining the list in the dial plan is all that
bad, actually. Dump it in its own context and file...
If that isn't convenient enough I'd go for the Asterisk database next.
Also on the option list is private e164/enum or an SQL database.
/Benny
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Re: [asterisk-users] Moving from DSL to T1
On Mon, 2010-09-13 at 00:32 -0700, Kevin Keane wrote: Latency also is the reason VoIP does not work at all over satellite connections even though they tend to have plenty of bandwidth. Please define does not work at all over satellite ??? Sure, it is not studio HIFI quality, but is th same quality as you get from official commercial telco providers. We still have voip over S-band and X-band satelites running NOW between NL and afghanistan. All the people are more than satisfied. ** Should have been more specific. I was talking about Internet over satellite in the USA. I believe those are geostationary TV satellites. I am not familiar with S-band and X-band, but assume they are in lower orbit. That would explain how it can work for you. No these are also geo-stationary (same altitude, so same delay), commercial and military satelites, -- _ -- 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] doing dnsmgr_lookup
On Mon, Sep 13, 2010 at 2:53 PM, Jonas Kellens jonas.kell...@telenet.be wrote: anyone on this list knows how to turn these messages off please ?! *CLI core set verbose 0 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving from DSL to T1
On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl wrote: No these are also geo-stationary (same altitude, so same delay), commercial and military satelites, Yes, exactly. Geostationary satellites have been used for telephone for ages (and are still used for remote areas - they have advantages over the disintegrating constellations such as iridium - namely predictability). As for consumer (home) grade satellite internet service, it's pretty low quality. But if you have money, you can have just as good of service as the telcos enjoy for TDM voice over them (even with VoIP). I know several organizations using them (but they are paying more than the $100 or so a month as is typical for a home user - a lot more). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 and fax
Hello! I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. When i try to receive fax I get: [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel 'SIP/crocus-ua-0004' refused to negotiate T.38 [Sep 13 00:46:02] WARNING[3283]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Sep 13 00:46:02] WARNING[3283]: app_fax.c:817 transmit: Transmission error I definitely know that this peer supports T.38 because it works on Lynksys PAP2T. Dialplan is such: answer() wait(6) ReceiveFAX(/var/spool/asterisk/test.tif) Am I doing something wrong here? Thanks! -- Stas Korsei On Thu, Sep 9, 2010 at 12:17 AM, David Backeberg dbackeb...@gmail.comwrote: On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei kor...@rinogo.com wrote: Can you recommend any specific solution to this problem or way to install app_fax? Not without specific debugging about what problems you're seeing. You get a lot of information when faxes succeed or fail. Try a fax and paste in the debug. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 and fax
On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei kor...@rinogo.com wrote: Hello! I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. When i try to receive fax I get: [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel 'SIP/crocus-ua-0004' refused to negotiate T.38 [Sep 13 00:46:02] WARNING[3283]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Sep 13 00:46:02] WARNING[3283]: app_fax.c:817 transmit: Transmission error I definitely know that this peer supports T.38 because it works on Lynksys PAP2T. There are lots of devices that 'support' T.38, but the problem is that they 'support' it differently. If you want to have fun, read the release notes for a Cisco voice IOS, and grep for the word T.38 to see the long list of known broken situations. Just because it's 'supported', doesn't mean it works. Internet Explorer 'supports' html, but good luck getting it to act like a standards-compliant web browser. Try turning off the T.38 and do analog passthrough, or try using two T.38 PAP2Ts. Or even better, don't use fax if you can avoid it. -- _ -- 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] Force ip disconnect after register?
On 09/13/2010 10:22 AM, Bryant Zimmerman wrote: Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box with reg attempts as long as they maintain their initial connection or I reset the state tables on the firewall. This is very bad. Is there a way to force the connection to drop and reconnect after let's say 50 attempts. Reconfigure your firewall to inspect every packet against the rules, instead of shortcutting 'open connections'; this takes more CPU on your firewall, but allows you to change the rules and drop existing connections. Alternatively, depending on how you've built your firewall, you can insert the 'drop all packets from X.X.X.X' *before* any rules that allow packets from existing connections. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving from DSL to T1
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Monday, September 13, 2010 12:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Moving from DSL to T1 On Mon, 2010-09-13 at 00:32 -0700, Kevin Keane wrote: Latency also is the reason VoIP does not work at all over satellite connections even though they tend to have plenty of bandwidth. Please define does not work at all over satellite ??? Sure, it is not studio HIFI quality, but is th same quality as you get from official commercial telco providers. We still have voip over S-band and X-band satelites running NOW between NL and afghanistan. All the people are more than satisfied. ** Should have been more specific. I was talking about Internet over satellite in the USA. I believe those are geostationary TV satellites. I am not familiar with S-band and X-band, but assume they are in lower orbit. That would explain how it can work for you. No these are also geo-stationary (same altitude, so same delay), commercial and military satelites, ** In that case, my guess is that they have a dedicated channel for the voice, maybe even some kind of clocking mechanism. Some T-1 lines here in the USA also have that (one more reason why T-1 works better than DSL/Cable for VoIP). The consumer internet satellite services just mix all kind of Internet traffic, so one packet may have a very low latency while the next one may have a much higher latency, or get lost altogether. Another thing about the consumer satellites is that they are probably optimized for TCP rather than UDP. For TCP, they are using huge retransmission window sizes. That allows large chunks of data to arrive without waiting for confirmation, and the satellite can organize the data into a stream. With UDP, each packet basically stands on its own. Just a guess about another area where these two could be different. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 and fax
On 09/14/2010 04:33 AM, Stanislav Korsei wrote: Hello! I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. When i try to receive fax I get: Why install 0.0.5? Its ancient. the world has moved on. [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel 'SIP/crocus-ua-0004' refused to negotiate T.38 [Sep 13 00:46:02] WARNING[3283]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Sep 13 00:46:02] WARNING[3283]: app_fax.c:817 transmit: Transmission error I definitely know that this peer supports T.38 because it works on Lynksys PAP2T. The Linksys PAP2T does NOT support T.38, so this statement makes no sense. The Linksys SPA2102 and SPA3102 support T.38. The PAP2 and PAP2T do not. Dialplan is such: answer() wait(6) ReceiveFAX(/var/spool/asterisk/test.tif) Am I doing something wrong here? Apparently. Steve -- _ -- 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] Moving from DSL to T1
On 09/14/2010 04:23 AM, Joel Maslak wrote: On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl mailto:h...@a-domani.nl wrote: No these are also geo-stationary (same altitude, so same delay), commercial and military satelites, Yes, exactly. Geostationary satellites have been used for telephone for ages (and are still used for remote areas - they have advantages over the disintegrating constellations such as iridium - namely predictability). When geostationary satellites were the normal thing for intercontinental calls, the call was normally satellite one way and cable the other. Satellite both ways would have been cheaper, but the total round trip latency was go bad, it was hard to hold a proper conversation. As for consumer (home) grade satellite internet service, it's pretty low quality. But if you have money, you can have just as good of service as the telcos enjoy for TDM voice over them (even with VoIP). I know several organizations using them (but they are paying more than the $100 or so a month as is typical for a home user - a lot more). Steve -- _ -- 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] Which 1.6 subversion is Stable one?
Hi all, I would like to install asterisk as my home pbx, Anyone can suggest which sub version of 1.6 is stable? Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom dhcp boot
Your string and boot-option look good. In the SonicWall config, its a two step process: - create the new boot option under DHCP Server menu Advanced button Add Option - assign it to your lease scope under DHCP Server menu DHCP Server Lease Scopes section Edit button Advanced tab DHCP Generic Options. Works on all my SonicWall TZ-200 with SonicOS Enhanced 5.6.2.0-2o.. but I recall older versions being good too. Bests, Sebastien On 2010-09-10, at 7:35 PM, colin mcdermott wrote: Hi all I have a few Polycom 331's but after following allot of advice I can't get them to provision from a dhcp boot server. We have a sonicwall router in place. I can press setup and set the FTP boot server to my * box. From there th phones boot fine. But I cannot get them to autoprovision. I have tried dhcp option 66=ipaddres. 66=FTP://PlcmSpIp:plcms...@192.168.1.1/ u ahve also tried options 129, 150, 160, etc. I realise that this is not an asterisk issue. But does anyone have any experience on this (particularly using sonicwall routers for Dhcp)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speech To Text on linux with asterisk
Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -- _ -- 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] Speech To Text on linux with asterisk
On Tue, Sep 14, 2010 at 1:01 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? ASR, yes. http://www.digium.com/en/products/software/lumenvox.php -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech To Text on linux with asterisk
Thanks Paul, i think still i have some problem to understand , i mean to say that i have 30 minutes audio file in WAV format and i wnat its text here are the scenario . - Call comes in - start recording - call remains for 30 minutes - stop recording - convert wav file audio to text. is this possible with lumenvox or any other engine. regards Dhaval On Tue, Sep 14, 2010 at 10:57 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Sep 14, 2010 at 1:01 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? ASR, yes. http://www.digium.com/en/products/software/lumenvox.php -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech To Text on linux with asterisk
It is simply not possible, though it might be in the distant future. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 1:50 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Thanks Paul, i think still i have some problem to understand , i mean to say that i have 30 minutes audio file in WAV format and i wnat its text here are the scenario . - Call comes in - start recording - call remains for 30 minutes - stop recording - convert wav file audio to text. is this possible with lumenvox or any other engine. regards Dhaval On Tue, Sep 14, 2010 at 10:57 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, ... -- _ -- 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
