Re: [asterisk-users] suggestions for low-power, small form-factor box with PCI and PCIe slots?
You could base your box on a motherboard with an onboard CPU (like Intel Atom). The disadvantage of these boards is that they usually come only with a single PCI or PCIe slot. There are industrial boards with different options, but they are rather expensive. The idle power of Sandy/Ivy Bridge systems is not too much different, so a H61 board could be a good choice, especially since you can use consumer components. I use server boards with C20x chipsets and a typical machine with 1 or 2 TDM cards has a total power consumption of about 35-40W in idle state. Since you probably deal only with at most a handful channels, a Celeron G465 should more than sufficient. I also prefer small SSDs instead of a hard disk. Let's say you have a 40 GB disk and the operating system plus additional software occupies 3-5 GB, then 90% is free which should result in a long life time since there is plenty of room for wear leveling optimizations. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Using PauseMonitor with MixMonitor
On 12 July 2013 16:36, Richard Mudgett rmudg...@digium.com wrote: On Fri, Jul 12, 2013 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature 'testfeature' == Mapping Feature 'testfeature' to app 'Playback(tt-monkeys)' with code '#9' == Registered Feature 'pauseMonitor' == Mapping Feature 'pauseMonitor' to app 'pauseMonitor()' with code '#00' == Registered Feature 'unpauseMonitor' == Mapping Feature 'unpauseMonitor' to app 'UnpauseMonitor()' with code '#01' -- Feature Found: pauseMonitor exten: pauseMonitor -- Executing [h@x:1] System(SIP/xxx.xxx.x.xxx-000c, php agi-bin/process-call.php 0 2013-07-12 15:09:30 inbound) == MixMonitor close filestream == End MixMonitor Recording SIP/213.166.5.185-000c I've tried executing it on self and peer with the same result. Any thoughts? The PauseMonitor and UnpauseMonitor applications work with the Monitor application not MixMonitor. If a Monitor is not active on the channel then the PauseMonitor and UnpauseMonitor applications hangup the channel. Richard Thanks for that. Is there any way to pause and unpause MixMonitor then? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Dongle or extra channel and sip SMS
On 15/7/13 3:00 am, bilal ghayyad wrote: I need to be able to send SMS messages for campaign or for specific users, also I need to be able to receive SMS messages and do automatic reply. In my experience, SMS is something best done out of Asterisk. That's not to say that Asterisk can't do it, of course, just that there are providers out there who can give you a nice friendly API for easy integration into your application. This is especially true if you need to send *lots* of messages in a short space of time: simply adding a single mobile device with a single SIM isn't going to cut it - you're going to need a bunch of them, at least. All of those will likely have different numbers, so you're going to have to handle that for receiving messages. Then you have to consider that some networks will charge more to send messages to numbers on the same network vs. a different network, so you might have to separate out your numbers into networks (easy if they've never been ported; more tricky if they have). Based on past projects (in the UK), the cost of multiple SIM contracts, the necessary hardware to connect them, development time, etc., is usually more than the cost of paying a third party with a suitable API x per message to deliver them on your behalf. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] External Recording Server for Asterisk Voicemail
Hello All, I'm planning to use Asterisk only for voicemail Application and Recording will be done at different server. When user changing his personal greeting or leaving voicemail Call need to throw to external Voicemnail recording server over SIP til the time recording complete. While throwing Cal from Asterisk to application box i have to use SIP request which having some string in R-URI. Please let me know is this possible with configuration example. Regards Amit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI timeouts
Hi, 1. Java process sends a request (e.g., add member to queue) Do you see the TCP ACK coming back from Asterisk? Yes, I do. During the quiet period while you're waiting for the response, do you receive events over that AMI connection? Yes. Are there other actions that you're attempting to execute? In the particular case I'm looking at right now, yes (there's a QueuePause action followed closely by an Originate action). Is there any consistency as to which commands are getting delayed? Here's a breakdown from the last two weeks: Timeout Total % Command ~~~ ~ ~ ~~~ 178 74782.38 Command 5 18700.26 DBDel 804 135495.93 QueueAdd 2894 556215.20 QueuePause 660 138564.76 QueueRemove So it appears that most of the delays are from the queue module, which is understandable, because that's doing most of the work in our set-up. There are any number of reasons why the response would be delayed, but the 25 seconds delay you're seeing is excessive for any of the reasons I can think of. It turns out the timeout in the Java app is set to only 3 seconds, not 5, like I said in my previous email. What would be a reasonable delay time? In the case I'm looking at right now, the longest I can see is 7.2s. Looking in the Java app logs, I can see it occasionally (166 times over the last two weeks) timing out after five retries, which means it failed to get a response to any of the retries within three seconds. Packet loss could cause delays in getting responses, but usually not for the lengths of times you're talking about. There's nothing in the packet capture to indicate packet loss. Perhaps I should mention another issue that we've seen (and worked around) previously. Our Asterisk uses ODBC to talk to an Oracle database for realtime peers, for func_odbc and, back then, for CEL. The issue was that when there was a job running against the database which caused it to slow down, Asterisk dropped calls with the message no reply to our critical packet. As soon as we changed the database job to run at night (when the call centre is closed), this problem went away. It feels like Asterisk was stuck waiting for the database and missed the critical packets when they were, in fact, there. Thanks for your help! Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI timeouts
What would be a reasonable delay time? In the case I'm looking at right now, the longest I can see is 7.2s. Looking in the Java app logs, I can see it occasionally (166 times over the last two weeks) timing out after five retries, which means it failed to get a response to any of the retries within three seconds. The view of the person who wrote the Java app is that three seconds is a long time: if an agent presses the pause button in their GUI, during those three seconds, they may receive a call. He says that when this system first went live, his app received responses within milliseconds. I'm assuming this is to do with increasing call volume, though it could be something else. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI timeouts
When you have many calls, there are usually (read/write=all) a lot of RTP, RTCP, and VarSet events. This might slow down things, but whether they occur or not depends on your configuration. This might be another thing to look at. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI timeouts
On Mon, Jul 15, 2013 at 7:59 AM, jg webaccou...@jgoettgens.de wrote: When you have many calls, there are usually (read/write=all) a lot of RTP, RTCP, and VarSet events. This might slow down things, but whether they occur or not depends on your configuration. This might be another thing to look at. When you execute an Originate action, are you doing so synchronously or asynchronously? A synchronous Originate performs the full outbound dial operation on the thread servicing the AMI request. Since each session in AMI gets its own thread that services both actions and events, a synchronous Originate can block that session from receiving events until it completes. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk offline compiling with get_mp3_source.sh
I need to make a Asterisk 18.0's offline compiling, SVN mp3 support
sources downloading does't particulary works cause my asterisk is in an
isolated network with NO network access whatsoever, I ve read this thread (
http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html) but
I 'm not understading one thing, because I download the file and run the
script but there is no asterisk-contrib-mp3.tar.gz in my tmp folder
--- contrib/scripts/get_mp3_source.sh.orig 2013-06-04 12:41:08.222602824
+0200
+++ contrib/scripts/get_mp3_source.sh 2013-06-04 12:40:45.218602846 +0200
@@ -9,6 +9,15 @@
exit 1
fi
+LOCAL_COPY=/tmp/asterisk-contrib-mp3.tar.gz
+if [ -f ${LOCAL_COPY} ]; then
+echo ***
+echo Found ${LOCAL_COPY} - unpacking it, not downloading
+echo ***
+tar xzf ${LOCAL_COPY}
+exit 0
+fi
+
svn export http://svn.digium.com/svn/thirdparty/mp3/trunk addons/mp3 $@
exit 0
and i don't know what to do with the mpglib file
asterisk (1:1.8.13.1~dfsg-3) mpglib Summary
addons/mp3/MPGLIB_README | 39
addons/mp3/MPGLIB_TODO |2
addons/mp3/Makefile | 24
addons/mp3/README|1
addons/mp3/common.c | 267 ++
addons/mp3/dct64_i386.c | 335 +++
addons/mp3/decode_i386.c | 153 +++
addons/mp3/decode_ntom.c | 219 +
addons/mp3/huffman.h | 332 +++
addons/mp3/interface.c | 323 +++
addons/mp3/layer3.c | 2029 +++
addons/mp3/mpg123.h | 132 +++
addons/mp3/mpglib.h | 75 +
addons/mp3/tabinit.c | 81 +
14 files changed, 4012 insertions(+)
--
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Re: [asterisk-users] Asterisk offline compiling with get_mp3_source.sh
Hi
You must copy the directory mp3, to the addons directory, where you put the
source asterisk code, and recompile it, again.
Kind Regards
On Mon, Jul 15, 2013 at 9:25 AM, leonardo collantes leonardo07...@gmail.com
wrote:
I need to make a Asterisk 18.0's offline compiling, SVN mp3 support
sources downloading does't particulary works cause my asterisk is in an
isolated network with NO network access whatsoever, I ve read this thread (
http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html)
but I 'm not understading one thing, because I download the file and run
the script but there is no asterisk-contrib-mp3.tar.gz in my tmp folder
--- contrib/scripts/get_mp3_source.sh.orig2013-06-04 12:41:08.222602824
+0200
+++ contrib/scripts/get_mp3_source.sh 2013-06-04 12:40:45.218602846 +0200
@@ -9,6 +9,15 @@
exit 1
fi
+LOCAL_COPY=/tmp/asterisk-contrib-mp3.tar.gz
+if [ -f ${LOCAL_COPY} ]; then
+echo ***
+echo Found ${LOCAL_COPY} - unpacking it, not downloading
+echo ***
+tar xzf ${LOCAL_COPY}
+exit 0
+fi
+
svn export http://svn.digium.com/svn/thirdparty/mp3/trunk addons/mp3 $@
exit 0
and i don't know what to do with the mpglib file
asterisk (1:1.8.13.1~dfsg-3) mpglib Summary
addons/mp3/MPGLIB_README | 39
addons/mp3/MPGLIB_TODO |2
addons/mp3/Makefile | 24
addons/mp3/README|1
addons/mp3/common.c | 267 ++
addons/mp3/dct64_i386.c | 335 +++
addons/mp3/decode_i386.c | 153 +++
addons/mp3/decode_ntom.c | 219 +
addons/mp3/huffman.h | 332 +++
addons/mp3/interface.c | 323 +++
addons/mp3/layer3.c | 2029
+++
addons/mp3/mpg123.h | 132 +++
addons/mp3/mpglib.h | 75 +
addons/mp3/tabinit.c | 81 +
14 files changed, 4012 insertions(+)
--
_
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--
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Re: [asterisk-users] AMI timeouts
I guess this was a question for Alexander. As far as I am concerned, I never had such a load that slowed down AMI event processing (responses within at most 1/10 of a second), but for future tests I should probably set up a real torture test. For a robust PBX application, it would make sense to have an event thread and separate action threads that handle actions and associated events. I think I see your point, Matt. Thanks for all the fish. When you execute an Originate action, are you doing so synchronously or asynchronously? A synchronous Originate performs the full outbound dial operation on the thread servicing the AMI request. Since each session in AMI gets its own thread that services both actions and events, a synchronous Originate can block that session from receiving events until it completes. Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] ignore 183 session progress in parallel call scenarios
Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know which early media audio stream to pass back to the caller in a parallel call scenario (as in this case several endpoint may indicate session progress all at the same time). The question is why is asterisk still sending 183 session progress back to the caller if no audio is to be bridged before the 200 OK anyway? If 183 are not passed back to the caller, then at least a 180 Ringing that may come from another endpoint will cause the calling endpoint to generate local ringback. This won't happen if the caller has received a 183 already. So it's a bit of a race condition as well - if the first endpoint to reply sends a 183 session progress this means the caller will not hear any ringback even if some of the other endpoints are sending back 180 Ringing. The question is can I somehow block 183 messages from being passed back to the calling endpoint when dialing several destinations in parallel? I don't see a point (please correct me if I'm wrong) to pass only the 183 SIP message back to the caller without the corresponding RTP stream, so it may be much better to actually ignore it when dealing with parallel call scenarios (bug?). BR, Hristo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] ignore 183 session progress in parallel call scenarios
On 15 July 2013 15:14, Hristo Trendev dist.li...@gmail.com wrote: Hi, I am using asterisk 1.8.22 and have a problem when calling in parallel several SIP endpoints and I am not sure how to resolve it. In this case Asterisk will not bridge any audio to the caller before the 200 OK. Which means any progress announcements, including remotely generated ringback, are not passed back to the caller. This behavior is completely correct, because there is no way to know which early media audio stream to pass back to the caller in a parallel call scenario (as in this case several endpoint may indicate session progress all at the same time). The question is why is asterisk still sending 183 session progress back to the caller if no audio is to be bridged before the 200 OK anyway? If 183 are not passed back to the caller, then at least a 180 Ringing that may come from another endpoint will cause the calling endpoint to generate local ringback. This won't happen if the caller has received a 183 already. So it's a bit of a race condition as well - if the first endpoint to reply sends a 183 session progress this means the caller will not hear any ringback even if some of the other endpoints are sending back 180 Ringing. The question is can I somehow block 183 messages from being passed back to the calling endpoint when dialing several destinations in parallel? I don't see a point (please correct me if I'm wrong) to pass only the 183 SIP message back to the caller without the corresponding RTP stream, so it may be much better to actually ignore it when dealing with parallel call scenarios (bug?). BR, Hristo -- _ I think you need to look into early media and also this application https://wiki.asterisk.org/wiki/display/AST/Application_Progress -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 offline compiling with get_mp3_source.sh
On Monday 15 July 2013, leonardo collantes wrote: I need to make a Asterisk 18.0's offline compiling, SVN mp3 support sources downloading does't particulary works cause my asterisk is in an isolated network with NO network access whatsoever, I ve read this thread ( http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html) but I 'm not understading one thing, because I download the file and run the script but there is no asterisk-contrib-mp3.tar.gz in my tmp folder You need to download asterisk-contrib-mp3.tar.gz from somewhere on a machine with Internet access, save it on a USB stick and save the file in /tmp/ . Alternatively, you could go to a connected machine, and run $ mkdir /tmp/mp3 $ svn export http://svn.digium.com/svn/thirdparty/mp3/trunk /tmp/mp3 $ cp -a /tmp/mp3 /media/usb0/ When finished, cp -a the mp3 folder to /usr/src/asterisk/addons/ on the Asterisk machine. If all else fails, you can set up a temporary Internet connection by plugging in a mobile phone and tethering to it! If there is wi-fi on site, you won't even run up any data charges. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] PoE L3 Switches
On Sun, July 14, 2013 18:36, bilal ghayyad wrote: Hello; Anyone used PoE L2 network switches other than cisco and recommend this for us? We need it to be stable and costly effective. Regards Bilal We use multiple Cisco SG100D-08P eight-port POE unmanaged switches with each located close to the end users, often on their desks. These devices are relatively cheap, seem to work well, and provide four POE ports plus four standard Ethernet ports. We find that this arrangement works for two or three employee work centres on each switch. So, it is a Cisco solution which you deprecate. However, it does work and it is both inexpensive and flexible. If distributing power from a central location over Ethernet is an absolute requirement then these probably are not what you want. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] PoE L3 Switches
Unless it runs IOS, I don't think most of us would consider that box a Cisco Likely it is a Cisco branded switch with Linksys hardware, i.e. consumer grade stuff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James B. Byrne Sent: Monday, July 15, 2013 11:35 AM To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE L3 Switches On Sun, July 14, 2013 18:36, bilal ghayyad wrote: Hello; Anyone used PoE L2 network switches other than cisco and recommend this for us? We need it to be stable and costly effective. Regards Bilal We use multiple Cisco SG100D-08P eight-port POE unmanaged switches with each located close to the end users, often on their desks. These devices are relatively cheap, seem to work well, and provide four POE ports plus four standard Ethernet ports. We find that this arrangement works for two or three employee work centres on each switch. So, it is a Cisco solution which you deprecate. However, it does work and it is both inexpensive and flexible. If distributing power from a central location over Ethernet is an absolute requirement then these probably are not what you want. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] PoE L3 Switches
Yes, that is one of the former Linksys branded switches, now labeled as Cisco Small Business. I've used quite a few in small business installs. They work well, are unmanaged, and inexpensive. Jerome On Mon, Jul 15, 2013 at 10:40 AM, Eric Wieling ewiel...@nyigc.com wrote: Unless it runs IOS, I don't think most of us would consider that box a Cisco Likely it is a Cisco branded switch with Linksys hardware, i.e. consumer grade stuff. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of James B. Byrne Sent: Monday, July 15, 2013 11:35 AM To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PoE L3 Switches On Sun, July 14, 2013 18:36, bilal ghayyad wrote: Hello; Anyone used PoE L2 network switches other than cisco and recommend this for us? We need it to be stable and costly effective. Regards Bilal We use multiple Cisco SG100D-08P eight-port POE unmanaged switches with each located close to the end users, often on their desks. These devices are relatively cheap, seem to work well, and provide four POE ports plus four standard Ethernet ports. We find that this arrangement works for two or three employee work centres on each switch. So, it is a Cisco solution which you deprecate. However, it does work and it is both inexpensive and flexible. If distributing power from a central location over Ethernet is an absolute requirement then these probably are not what you want. -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PoE L3 Switches
On Mon, Jul 15, 2013 at 8:40 AM, Eric Wieling ewiel...@nyigc.com wrote: Unless it runs IOS, I don't think most of us would consider that box a Cisco Likely it is a Cisco branded switch with Linksys hardware, i.e. consumer grade stuff. They work well in small business. They have a command line that looks and feels like IOS, though I have no idea if it is or not. Note that there is a confirmed bug with LLDP and auto-VLAN on the SF/SG switches and I haven't heard that they fixed it. If you have phones with CDP, or manually provision VLANs, or don't use VLANs, then no problem. They work great and are reliable. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] PoE L3 Switches
I have a few TP-LINK TL-SF1008P and D-Link DGS-1008P running in office environments, but I prefer the D-Link DGS-1210-10P (with fan) at a central location if the cable lengths permit it. A couple of years ago I had 2 broken Netgear devices that ran about half a year, but I cannot say anything about the current models. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Dongle or extra channel and sip SMS
Something to check out: http://www.kickstarter.com/projects/smush/smart-sms-texting-for-everyone-the-smushbox I'm not affiliated with them at all, but have done business with the company on other things and have always been happy. On Mon, Jul 15, 2013 at 1:57 AM, Chris Bagnall aster...@lists.minotaur.ccwrote: On 15/7/13 3:00 am, bilal ghayyad wrote: I need to be able to send SMS messages for campaign or for specific users, also I need to be able to receive SMS messages and do automatic reply. In my experience, SMS is something best done out of Asterisk. That's not to say that Asterisk can't do it, of course, just that there are providers out there who can give you a nice friendly API for easy integration into your application. This is especially true if you need to send *lots* of messages in a short space of time: simply adding a single mobile device with a single SIM isn't going to cut it - you're going to need a bunch of them, at least. All of those will likely have different numbers, so you're going to have to handle that for receiving messages. Then you have to consider that some networks will charge more to send messages to numbers on the same network vs. a different network, so you might have to separate out your numbers into networks (easy if they've never been ported; more tricky if they have). Based on past projects (in the UK), the cost of multiple SIM contracts, the necessary hardware to connect them, development time, etc., is usually more than the cost of paying a third party with a suitable API x per message to deliver them on your behalf. Kind regards, Chris -- This email is made from 100% recycled electrons -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Dongle or extra channel and sip SMS
On Monday 15 July 2013, bilal ghayyad wrote: Hello; I need to be able to send SMS messages for campaign or for specific users, also I need to be able to receive SMS messages and do automatic reply. Do I have to use dongle or extra channel? What is the difference? Also, I read that there is SMS through sip, how this work and what is the difference between the sip SMS and gsm sip? If I need to send sip SMS, how destination will receive it? What is required for destination phone to receive this sip SMS? We use OpenVox G400P cards (PCI; newer motherboards probably will require G400E which is essentially the same card but with PCI Express interface) for our SMS and mobile telephony needs. Just slot in up to 4 SIM cards, and install chan_extra. Incoming text messages trigger an extension in your dialplan with some channel variables set, so you can respond to the message (and also are recorded in a logfile). Outgoing messages can be sent using the Asterisk CLI. In principle, anything that you can call from a dialplan can be initiated by an incoming message, and anything that can execute a system command can initiate an outgoing message. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] ignore 183 session progress in parallel call scenarios
I think I have found the answer to my questions in the source code of Dial:
case AST_CONTROL_PROGRESS:
ast_verb(3, %s is making progress passing it to %s\n,
ast_channel_name(c), ast_channel_name(in));
/* Setup early media if appropriate */
if (single !caller_entertained
CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
if (single || (!single !pa-sentringing)) {
ast_indicate(in, AST_CONTROL_PROGRESS);
}
}
.
.
Asterisk will attempt to bridge the media only for the case of a
single outgoing channel, but at the same time it will happily forward
progress messages for parallel calls: (!single !pa-sentringing) as
long as no 180 Ringing message was sent out to the caller yet. The
questions still remains if this should be reported as bug or if there
is indeed a use case when sending 183 progress message, without
actually bridging the media stream is desired.
On Mon, Jul 15, 2013 at 4:14 PM, Hristo Trendev dist.li...@gmail.comwrote:
Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means any progress announcements, including remotely generated ringback,
are not passed back to the caller.
This behavior is completely correct, because there is no way to know which
early media audio stream to pass back to the caller in a parallel call
scenario (as in this case several endpoint may indicate session progress
all at the same time).
The question is why is asterisk still sending 183 session progress back to
the caller if no audio is to be bridged before the 200 OK anyway? If 183
are not passed back to the caller, then at least a 180 Ringing that may
come from another endpoint will cause the calling endpoint to generate
local ringback. This won't happen if the caller has received a 183 already.
So it's a bit of a race condition as well - if the first endpoint to reply
sends a 183 session progress this means the caller will not hear any
ringback even if some of the other endpoints are sending back 180 Ringing.
The question is can I somehow block 183 messages from being passed back to
the calling endpoint when dialing several destinations in parallel? I don't
see a point (please correct me if I'm wrong) to pass only the 183 SIP
message back to the caller without the corresponding RTP stream, so it may
be much better to actually ignore it when dealing with parallel call
scenarios (bug?).
BR,
Hristo
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[asterisk-users] Asterisk 1.8.23.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.23.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix a memory copying bug in slinfactory which was causing mixmonitor issues. (Closes issue ASTERISK-21799. Reported by Michael Walton) * --- IAX2: fix race condition with nativebridge transfers. (Closes issue ASTERISK-21409. Reported by alecdavis) * --- Fix crash in chan_sip when a core initiated op occurs at the same time as a BYE (Closes issue ASTERISK-20225. Reported by Jeff Hoppe) * --- Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit (Closes issue ASTERISK-21246. Reported by Peter Katzmann) * --- chan_sip: Session-Expires: Set timer to correctly expire at (~2/3) of the interval when not the refresher (Closes issue ASTERISK-21742. Reported by alecdavis) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.23.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.5.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix Segfault In app_queue When persistentmembers Is Enabled And Using Realtime (Closes issue ASTERISK-21738. Reported by JoshE) * --- IAX2: fix race condition with nativebridge transfers. (Closes issue ASTERISK-21409. Reported by alecdavis) * --- Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit (Closes issue ASTERISK-21246. Reported by Peter Katzmann) * --- Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX (Closes issue ASTERISK-21374. Reported by Michael L. Young) * --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent out after retries fail (Closes issue ASTERISK-21677. Reported by Dan Martens) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Jitter buffer on write side of channel
How does one do this? We have a particular SIP phone that needs a large jitterbuffer, but all I can see is how to put it on the *read* side of the channel. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] External Recording Server for Asterisk Voicemail
If you want to store in external, why can't you have a NAS device and mount to Asterisk server, let the mounted be a part in asterisk.conf, so that voicemail will get recorded in external server... Will it makes sense... ! Thanks. On Mon, Jul 15, 2013 at 4:19 PM, Amit Salunkhe amitsalunkh...@gmail.comwrote: Hello All, I'm planning to use Asterisk only for voicemail Application and Recording will be done at different server. When user changing his personal greeting or leaving voicemail Call need to throw to external Voicemnail recording server over SIP til the time recording complete. While throwing Cal from Asterisk to application box i have to use SIP request which having some string in R-URI. Please let me know is this possible with configuration example. Regards Amit -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Jitter buffer on write side of channel
On Jul 15, 2013, at 3:35 PM, Richard Kenner ken...@gnat.com wrote: How does one do this? We have a particular SIP phone that needs a large jitterbuffer, but all I can see is how to put it on the *read* side of the channel. At the risk of being a little tangential, what is a write-side jitterbuffer? smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Jitter buffer on write side of channel
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Behrens Sent: Monday, July 15, 2013 6:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Jitter buffer on write side of channel On Jul 15, 2013, at 3:35 PM, Richard Kenner ken...@gnat.com wrote: How does one do this? We have a particular SIP phone that needs a large jitterbuffer, but all I can see is how to put it on the *read* side of the channel. At the risk of being a little tangential, what is a write-side jitterbuffer? It is a kind of time travel where you dejitter the packets before the packets have jitter. The correct place to remove jitter is on the receiving endpoint.The jitterbuffer settings are most useful for things like voicemail, recordings, meetme, etc where Asterisk is endpoint. Even then, only the received audio needs to be dejittered. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Microsoft CRM Integration
Hi All, I'm hoping someone can recommend a method to integrate Microsoft CRM with Asterisk. Preferably an open source product otherwise a commercial product. Regards David Klaverstyn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
