Re: [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov d...@belkam.com wrote:
Hello!
Just installed asterisk 13.2.0 and see many such messages in log, I see them
in console during calls, really something like this:
-- Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6,
SIP/6166@asterisk) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/6166@asterisk
0x7fa9d4007660 -- Probation passed - setting RTP source address to
10.18.0.19:26052
-- SIP/asterisk-000c is making progress passing it to
OOH323/kanbaikal-6
-- SIP/asterisk-000c is ringing
0x7fa9d4007660 -- Probation passed - setting RTP source address to
10.18.0.19:26052
0x7fa9d800d610 -- Probation passed - setting RTP source address to
192.168.166.2:2080
[Mar 5 11:13:14] ERROR[3526]: json.c:704 ast_json_vpack: Error building
JSON from '{s: s, s: s}': Invalid UTF-8 string.
[Mar 5 11:13:14] ERROR[3526]: json.c:704 ast_json_vpack: Error building
JSON from '{s: s, s: s}': Invalid UTF-8 string.
-- SIP/asterisk-000c answered OOH323/kanbaikal-6
-- Channel OOH323/kanbaikal-6 joined 'simple_bridge' basic-bridge
d751932e-2e26-4671-8fcc-048b20156ec9
-- Channel SIP/asterisk-000c joined 'simple_bridge' basic-bridge
d751932e-2e26-4671-8fcc-048b20156ec9
0x7fa9d800d610 -- Probation passed - setting RTP source address to
192.168.166.55:3098
[Mar 5 11:13:19] ERROR[3526]: json.c:704 ast_json_vpack: Error building
JSON from '{s: s, s: s}': Invalid UTF-8 string.
[Mar 5 11:13:19] ERROR[3526]: json.c:704 ast_json_vpack: Error building
JSON from '{s: s, s: s}': Invalid UTF-8 string.
[Mar 5 11:13:24] ERROR[3526]: json.c:704 ast_json_vpack: Error building
JSON from '{s: s, s: s}': Invalid UTF-8 string.
[Mar 5 11:13:24] ERROR[3526]: json.c:704 ast_json_vpack: Error building
JSON from '{s: s, s: s}': Invalid UTF-8 string.
[Mar 5 11:13:29] ERROR[3526]: json.c:704 ast_json_vpack: Error building
JSON from '{s: s, s: s}': Invalid UTF-8 string.
This is call from H323, as I know avaya , chan_ooh323 from my side to
another asterisk SIP chan_sip on both sides.
Just because everything work OK, I , definitely, can comment out this error
message, but...
Could you give me any idea why this error can appear?
If you haven't create an issue on Jira, this is a bug.
--
Paul Belanger | PolyBeacon, Inc.
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Re: [asterisk-users] WebRTC phone
On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens jar...@mogl.com wrote: For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages: kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-carrierroute.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-cpl.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-debuginfo.x86_644.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-gzcompress.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-ims.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-ldap.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-mysql.x86_644.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-outbound.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-postgres.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-presence.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-python.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-sctp.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-snmpstats.x86_644.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-sqlite.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-tls.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-unixodbc.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-utils.x86_644.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-uuid.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-websocket.x86_644.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-xml.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-xmpp.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms Keep in mind that using Kamailio to bridge the signalling is only half of the equation. You must also bridge the media and so the rtpengine module allows Kamailio to interface with the rtpengine (https://github.com/sipwise/rtpengine) which does that half. In the provided example Kamailio.cfg there isn't any real hardening and it's pretty much purely used as a bridge that would front an Asterisk 1.8 server for webrtc capabilities (but not any other sip). It uses the dispatcher module to dispatch to the underlying asterisk so you will still need to add the Asterisk to the dispatcher config. +1 to everything here. We also do this and it works quiet well. Kudos. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Is Asterisk a Linux only system?
On Thu, Feb 12, 2015 at 11:40 AM, Matthew Jordan mjor...@digium.com wrote: On Thu, Feb 12, 2015 at 8:52 AM, D'Arcy J.M. Cain da...@vex.net wrote: On Thu, 12 Feb 2015 09:43:33 -0500 Ron Wheeler rwhee...@artifact-software.com wrote: Why not just bite the bullet and move to a supported Linux? If all I had was a phone switch that might be an option but this is just part of a multi-server system that needs to be able to move services back and forth so the underlying OS has to be the same for everything. Besides, I am a NetBSD developer and so I am also interested in making every package rock solid on it. - you can be assured that it works - updates are tested I would be willing to make a NetBSD machine (not my production server) available for running unit tests. Are there already unit tests in the distribution? Yes there are. In addition to unit tests, there are also the functional tests in the Asterisk Test Suite [1]. To enable them as well as set up Asterisk for the Test Suite: 1. Configure Asterisk for development mode: $ ./configure --enable-dev-mode 2. In menuselect, enable the TEST_FRAMEWORK Compiler Flag 3. Also in menuselect, enable the Test Modules. These provide the unit tests. 4. Build/install Asterisk 5. Run Asterisk 6. Execute the unit tests (or a subset thereof) using the CLI: *CLI test execute [category|all] Note that some unit tests require a particular configuration or certain subsystems to be enabled. You can examine the CI build agent scripts used for test runs here: http://svn.asterisk.org/svn/testsuite/bamboo/trunk/bin/ Specifically, the build-asterisk-only.sh script and run-asterisk-unittests.sh. Setting up [2] and running [3] the Asterisk Test Suite is documented on the wiki, and generally covers a lot more functionality than the unit tests. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation [2] https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suite [3] https://wiki.asterisk.org/wiki/display/AST/Running+the+Asterisk+Test+Suite It should be noted, we did have a FreeBSD and Ubuntu systems running the testsuite back in 2010. FreeBSD was donated to the project. I personally had a PowerPC system running asterisk / testsuite, on debian. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Question regarding custom announcements used by several Asterisk servers
On Fri, Feb 6, 2015 at 5:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote: On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Got a question regarding custom announcements in Asterisk. My goal is to allow my users record their own queue announcements and choose which announcements they want to use in each queue. I have several Asterisk servers and a Kamailio server which dispatches call traffic between the Asterisks. Question is, is it possible to have something like a NSF disk shared between several asterisk servers and store custom announcements there, where all Asterisks would use them? I expect to have to place the files under whatever I configure in asterisk.conf. Additionally, can I place the announcements in subfolders under that directory and in my realtime queue table use values something like '/subfldr/myannouncement'? Keep up the good work! cheers, Olli -- _ -- 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 Hi All of that is possible and is exactly what we do, both for customer sounds and for call recordings. Just make sure you have resilience in your shared storage device. Alternatively, you could use something like Puppet to deploy the files to all the servers. This is basically what we do, we use puppet to help distribute files to remote servers while still using app_queue. Shared network drive also works. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Java API - Up to date
On Tue, Jan 27, 2015 at 4:14 PM, symack sym...@gmail.com wrote: Hello Everyone, I am required to write a java program that will get our asterisk to: * Query the database for phone numbers * Loop through numbers and dial * Play message * Get dial pressed response - If 1 = Yes - If 2 = No - If 3 = Connect to Agent * AMD Capable * Disposition I am proficient with Java and found the Asterisk-Java API. My questions are: * What is the recommended API to use * Is Asterisk-Java API maintained by digium * Am I overlooking anything? Your help is greatly appreciated. There's many ways to accomplish this, many have been discussed on this mailing list. You are going to use the AMI to originate calls into asterisk. No, Asterisk-Java is not maintained my Digium. As for overlooking, likely, but you should be able to see anything you missed in your testing phase. You should be able to google Asterisk dialers to see some example that people have done. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 show queue-name vs queue log for calculating average hold time
On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show queue_name I get the following numbers: queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the average hold time before calls get answered by a queue members. However, if I calculate the average hold time from out queue log table using the following SQL select sum(data1)/ count(*) as ave_hold_time from queue_log where time DATE(NOW()) and queuename='queue_name' and event='CONNECT'; I get the vastly different figure of 92.4. So, is the queue show figure wrong due to a bug or am I making an incorrect assumption as to what it means? Thanks in advance Welcome to business logic embedded into app_queue. The issue with the queue show command rendering stats, is what timeframe are the stats aggregated over? IIRC, the calculations are using a moving average[1]. Opps, sent instead of pasting. Either way, your likely better off rendering the data using the raw sql info vs depending on CLI output. That's what we've done. [1] http://en.wikipedia.org/wiki/Moving_average -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 show queue-name vs queue log for calculating average hold time
On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show queue_name I get the following numbers: queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the average hold time before calls get answered by a queue members. However, if I calculate the average hold time from out queue log table using the following SQL select sum(data1)/ count(*) as ave_hold_time from queue_log where time DATE(NOW()) and queuename='queue_name' and event='CONNECT'; I get the vastly different figure of 92.4. So, is the queue show figure wrong due to a bug or am I making an incorrect assumption as to what it means? Thanks in advance Welcome to business logic embedded into app_queue. The issue with the queue show command rendering stats, is what timeframe are the stats aggregated over? IIRC, the calculations are using a moving average[1]. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Cannot get my first WebRTC experiment to work.
On Wed, Jan 28, 2015 at 8:27 AM, Antonio Gómez Soto antonio.gomez.s...@gmail.com wrote: Hi all, Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works. My network setup by the way: I am working from a cable modem, I created the test setup at digital ocean. From my laptop I also have a direct VPN connection to the asterisk server my laptop being 192.168.241.10 and asterisk being 192.168.241.30 I think something is wrong with the RTP address negotiation, but I have trouble interpreting the SDP wrt WebRTC and ICE. 1. asterisk seems to be telling sipml5 to send audio to it's public ip addres, but * sends to 192.168.241.10 2. the asterisk output does show RTP flows to chrome, but there's no sound from chrome. I hope someone can intersperse the output with comments? Pastebin the fill debug, you've delete an important piece of information. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] ${HASH(SIP_CAUSE,channel-name)}
On Thu, Oct 30, 2014 at 9:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I read on the wiki :
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}. Additionally make sure you're using the
destination channel, not the source channel.
But when I use this in my dialplan, this 'variable' is empty.
Dialplan :
exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})
CLI :
[Oct 30 14:48:03] -- Executing [h@pbx-routing:5]
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
Can anyone tell me how this should be used ?
sip.conf: storesipcause=yes
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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Re: [asterisk-users] Asterisk 12 - zombie processes
On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Mathew, When I run 'ps -ef|grep asterisk' the following processes are displayed: root 6861 1 0 Aug27 ?00:00:00 /bin/sh /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf asterisk 8062 6861 3 Oct27 ?00:44:56 /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c root 20776 2200 0 11:20 pts/200:00:33 tail -f asterisk.log asterisk 23076 8062 0 17:01 ?00:00:00 [asterisk] defunct asterisk 23897 8062 0 17:03 ?00:00:00 [asterisk] defunct also when I run top the same amount of zombie processes are displayed: Tasks: 185 total, 1 running, 182 sleeping, 0 stopped, 2 zombie Regarding the AGI - we are using AGI in order to run php scripts for external logic. I have printed the PIDs of the php scripts and none of them are related to the PID's of those zombie processes. Do you have any idea how to find out what are these processes? Yaron. Are you doing anything like: # asterisk -rx 'core show channels' via an external process? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 make voip client cannot use same username?
On Sun, Sep 28, 2014 at 11:51 AM, rafa alfurqan rafa.alfur...@gmail.com wrote: Hi All, I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too. what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) -- ok then in client 3, i used (1001) same username with client 1. when client 1 is connecting with client 2, my client 3 could make a call to with client 2 (1002) with the same username in client 1. how i could make the system, so i cannot use with 1 username in 2 different client before i make a call (when registering process in voip client), or at least my voip client cannot use same username if that username is connected with the other user? Since what you describe is a valid for SIP, you'll have to drop the packets at the network level (firewall). Or use the ACL system in asterisk to restrict it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 peer definition registration
On Sat, Aug 16, 2014 at 11:21 AM, Steve Ng steveng.1...@gmail.com wrote: Hi, I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my real-time, I would set the SIP credential based on what the user has provided. For example [name] type=peer defaultuser=USER_PROVIDED secret=USER_PROVIDED host=USER_PROVIDED When I reset Asterisk, Asterisk will attempt to register with the sip provider. And if there are sufficiently amount of records with invalid credentials, I'll get blocked by the SIP provider as they might think that I'm brute forcing. Just a question to check if there's any chance I could ask Asterisk not to register when I reset. Or is there any other possible solution for this? No, only reload after your ITSP brute force timer has expired. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
On Fri, Aug 15, 2014 at 10:41 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, After having thought this through a bit I have some thoughts I'd like to share. In this case where the rtp profile is RTP/AVP Asterisk accepts and handles the call normally. If a webrtc client calls a sip client, or even another webrtc client, rtpengine is needed to step in (in my setup most of the clients would indeed be webrtc, but some of them might be sip). I think it would be better to use RTP/SAVPF throughout the process if both clients are webrtc (or otherwise speak RTP/SAVPF), but currently there is no way to accomplish this? Is it possible to configure Asterisk to only accept the RTP/SAVPF profile, and send 488 to all others? If it's not possible to force Asterisk to ignore rtp profiles (thus allowing the sdp be handled by rtpengine entirely), I'd prefer to use RTP/SAVPF or RTP/SAVP in the communication between Kamailio and Asterisk servers and use rtpengine to bridge to RTP/AVP and RTP/AVPF only if the client cannot speak securely. I'd very much like to hear opinions and thoughts on these. Again, I'll only share my experiences, but we do the complete opposite. Traffic between kamailio and asterisk is only RTP/AVP since the version of asterisk we are using does not support RTP/SAVPF (1.8). However, if you want RTP/SAVPF then honestly, you should just completely remove rtpengine from the picture since newer version of asterisk support both RTP/AVP and RTP/SAVPF (asterisk 12+). What I think you should do is go back to the basics, and document everything you want to do. Right now you have too many pieces in the puzzle and making the setup complicated. Like I said before, this is a complex setup and you need to start some place. Here is a diagram of what we do. webrtc (RTP/SAVPF) - kamailio - rtpengine - asterisk (RTP/AVP) This way, only RTP/AVP is in the core of our network. Rtpengine is on the edge (where it belongs), proxing rtp traffic. And, for us, we keep RTP/SAVPF outside of asterisk since support for it has been recently added. I also believe there are some open issue with dtls + srtp too. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
On Fri, Aug 15, 2014 at 12:17 PM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Thanks Paul, I appreciate your thoughts. I understand your way, it's logical in your environment. I prefer to use LTS versions of Asterisk so I'm guessing what I want to do is not quite possible with Asterisk 11. I'd prefer my setup to work like this in different cases. webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- webrtc (rtp/savpf) sip (rtp/avp) -- kamailio -- rtpengine (rtp/savpf) -- asterisk -- kamailio -- rtpengine (rtp/avp) -- sip (rtp/avp) webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- rtpengine (rtp/avp) -- sip (rtp/avp) ... essentially, using RTP/AVP only when the client does not speak securely. It appears I'll have to try out the RTP/AVP way until there is an Asterisk that can accomplish this without having to use peer-specific settings. Down-side to this is that rtpengine needs resources from the server for webrtc clients even though both ends speak the same profile. It's not so complicated now that I know more on what Asterisk supports and how it handles the sdp, I just needed to learn by doing, testing and asking. I must be a bit ahead of my time for going for a RTP/SAVPF within my architecture, but using RTP/AVP is not such a bad option as srtp is on its way anyway in future Asterisk versions and the rtp flowing between Kamailio and users' networks are far more important than internal rtp traffic. Fair enough, I won't be able to help moving forward. We opted for only using RTP/AVF with asterisk because how new the code for RTP/AVPF and dtls-strp handling is. And since RTP/AVF has been around since the start, it is pretty stable. And this is the primary reason people are using rtpengine with asterisk to start. So, in your setup listed above, rtpengine is not needed, since newer versions of asterisk support both. Adding it in will just complicate your setup. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hi, Wow, thanks Paul, realizing the problem makes a lot of sense. So I setup Kamailio as a peer, but if I disable chan_sip module completely, I can't do it in sip.conf like I'd otherwise assume to do. I tried to rebuild Asterisk without chan_sip, but I guess that's not quite the way to go? Asterisk stopped sending back any sip messages so either there is a configuration means on how to do this or I'm doing something wrong with my current setup. My next thought was to compile Asterisk normally and set rtcachefriends to no, that did not work either, when dialing the cli stated: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) which I guess says Asterisk does not know where to send the message. The inner workings of Asterisk is a bit beyond me, if you don't mind giving advice on how to proceed I'd be most grateful. I think you are still mis-understanding me. I'll try to be clearer. From the POV of asterisk, you do still need chan_sip, however the only peer asterisk needs to be away of it Kamailio. All other peers will be stored within kamailio. This was the reason for my comment about realtime sip, you don't need it. Then, within kamailio, you'll need to invoke rtpengine using (rtpproxy-ng with kamailio 4.1) to rewrite the sdp for the invite to asterisk. You'll use the rtpproxy_offer and rtp_answer functions to remove ICE when calls originate from webrtc clients. Since you are not using a websocket in asterisk, it will just be a SIP over udp, the need for ICE and SAVPF is not needed. What you are trying to do is pretty complicated, it took me about 2 weeks to get everything setup properly. There is good information[1] on the web, you just need to google for it. [1] http://www.slideshare.net/crocodilertc/webrtc-websockets -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, Thank You Paul for your reply, The registrations in my setup are not duplicated, the 'secret' field in the realtime table is empty, which causes Asterisk to not authenticate requests from my Kamailio. Kamailio handles registrations, and also routes the traffic to Asterisk using dispatcher. Also, all peers have the Kamailio ip:port as outbound proxy so all traffic goes through Kamailio. That is your issue, stop using chan_sip with realtime (using data from kamailio). The only SIP peer asterisk should know of is kamailio, and your webrtc clients should be anonymous SIP users. This way, Asterisk doesn't even need to deal with websockets and RTP/SAVPF (this is what kamailio and rtpengine) is for. In your current setup, you are bypassing the functionality of rtpengine and not even leveraging it. Looks like version 11.11 works differently, I'll try to revert back to a previous version, and see if that works. I know at least the 'force_avp' field is new to 11.11 so it's safe to assume there's some difference between versions in rtp profile handling. It would be good to know how to handle this scenario in the new versions as well, I'll probably need to upgrade ahead anyway. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 12 on Debian Wheezy
On Tue, Aug 12, 2014 at 12:38 PM, Olivier oza.4...@gmail.com wrote: Hello, A couple of questions in relation with Asterisk 12 on Debian Wheezy. 1. Can paquet libpjproject-dev (from wheezy-backport) be installed as the sole binary to add PJSIP stack to Asterisk 12 (compiled from source) ? Yes, you should be able too. 2. When compiling PJPROJECT from source (see https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject?src=search)), where should PJSIP .so files be located in an appropriately managed Wheezy system ? In other words should I get the line bellow or something else ? libpjsua.so (libc6) = /usr/lib/libpjsua.so You will likely need to pass the pjproject directory to configure. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Letting rtp profiles be handled by rtpengine instead of Asterisk
On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen ohjelmistoarkkite...@gmail.com wrote: Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to configure Asterisk to ignore the rtp profile but allow calls to pass with either of those profiles (even though clients might answer with 488 which would be caught and handled by Kamailio and rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and the second goal is to be able to add peers to the db table with similar data, as in no different values based on what kind of client wants to register. I'd like to allow the user to register using which ever client they choose (in this case one of the 3 I mentioned). Previously I had problems like 'rejecting secure audio stream without encryption details', no audio or BYE messages sent immediately after call has begun etc, but according to sip.js documentation (http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf and force_avp affect the way Asterisk handles the rtp profiles and now my calls do work ok but I'd need to move the rtp profile handling to rtpengine. We are successfully using kamailio / rtpengine with websockets and asterisk 1.8. First question is why are you duplicating registrations within asterisk? Secondly, why are you using websockets in asterisk? Without knowing more about your use case, I'll tell you how we did it. Like I said, kamailio is responsible for our SIP/ws subscribers and registrations. Once within kamailio we simply dispatch traffic to asterisk via SIP/udp. RTP is handled by rtpengine (using rtproxy-ng) and that is basically it. No special configuration is needed for asterisk (in fact 1.8 has no support for RTP/SAVPF) so we rewrite SDP on 488. Then setup a kamailio peer and away you go. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Notification when queue member's phone rings
On Wed, Jul 2, 2014 at 4:11 PM, Mitch Claborn mitch...@claborn.net wrote: Short question: how to get control or notification (gosub, macro, AGI) when a queue member's phone starts ringing due to an incoming call from the queue. Backround: Our phone operators serve both an asterisk call queue and a queue for web chat support. I have a gosub on the queue that calls to our app server to mark the operator unavailable for web chat as soon as they answer an incoming queue call. Similarly, when a web chat is connected, it uses AMI to tell asterisk to take the operator out of the phone queue. The other day, one operator got a web chat that came in while her phone was ringing with a queue call, so that neither remove from queue operation was effective in time. If I could get notification when the phone starts ringing I can reduce the window of opportunity for that by several seconds. It's only happened once in 2 years that I know of, so may not be worth worrying about. AMI will raise the AgentCalled[1] event. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_AgentCalled -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Live Recording on the Storage Server?
On Thu, Apr 17, 2014 at 7:25 AM, binary dreamer dreamer.bin...@gmail.com wrote: hi. I would not do that due to network issues. My approach is to record everything locally and every hour or so to move everything to a storage. +1 save yourself the headache and do this. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 11 under VMware?
On Fri, Apr 4, 2014 at 1:30 PM, Carlos Chavez cur...@telecomabmex.com wrote: I have found Asterisk using only SIP is very responsive on virtual machines. We have used VMs for call center applications and for complex IVR solutions without problems. Obviously there is overhead running a VM so you can never expect a VM to perform as well as bare metal. Running a single VM on a server is a complete waste of resources, might as well run natively. Well, regardless of how many VMs you run on bare metal, you do get the benefit of the VM technology. Even if OP runs 1 VM on the box, he could leverage snapshots in VM ware for the purpose or migrating or back ups. I don't think it is a waste per say, just different requirements. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 CLI Banner
On Fri, Mar 28, 2014 at 2:39 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 28 Mar 2014, Richard Kenner wrote: And this certainly may vary from jurisdiction to jurisdiction. For a (quite dated at this point) discussion of this issue from a US perspective, see http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The publication (43 pages) is dated 1988. The DMCA (1998) and subsequent legislation may have changed the landscape. My (ignorant) opinion -- just don't. Is it worth the effort to research? Is it worth paying a lawyer to research it and give an opinion that may be worth nothing until it is examined in court? If you want to display something custom, how about a 'wrapper' script that displays a file using 'curl' before handing off to Asterisk -- easier to implement, easier to maintain, no legal BS to consider. Or can you express your creativity by fiddling with ASTERISK_PROMPT? If you really want to do it: 1) create a wrapper to asterisk -r 2) pipe the welcome message to /dev/null 3) ??? 4) profit you didn't modify Asterisk. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] asteriskdocs.org says 3rd ed. is latest
On Mon, Mar 24, 2014 at 8:54 AM, Matt Behrens m...@zigg.com wrote: I made myself look a little silly recently in a talk regarding asteriskdocs.org. I didn't realize the 4th ed. of the Definitive Guide was apparently actually out (http://shop.oreilly.com/product/0636920025894.do), because I went by asteriskdocs.org's claim that it was being worked on in OFPS (now retired, apparently.) Is the 4th ed. available to read online like the 3rd ed. was? Is someone on this list able to update asteriskdocs.org with current info regardless? I pinged Leif Madsen on this and he's updating the site now. Currently only the 3rd edition is published online. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Proxy
On Mon, Mar 24, 2014 at 6:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi people Just having a quick check to see if anyone is using any AMI proxies and which are the most popular. For our purposes it must be able to connect to multiple asterisk instances. All depends on the language you want to use. We used starpy for a while, but ended up rewriting our own version. Currently we're connecting AMI to a message bus and passing events across the bus. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] AppVoicemail overwrites voicemail.conf
On Fri, Mar 21, 2014 at 11:58 PM, Al lists asteris...@gmail.com wrote: looking more into this, looks like this is not a issue, its related to users changing voicemail password from handset, asterisk rewrites the file. Right, use passwordlocation = spool, create a secret.conf for each mailbox, now when a user changes their password, secret.conf gets updated not voicemail.conf. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] AppVoicemail overwrites voicemail.conf
On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote: We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not using realtime. anyway to prevent AppVoicemail ro auto generate files? passwordlocation = spooldir Read voicemail.conf about how to use it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need more meetme users -- hitting some limit
On Fri, Mar 21, 2014 at 11:53 AM, Steve Edwards asterisk@sedwards.com wrote: I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call file on box1 originates a call to box2 and then plays a 2 hour WAV file. The dialplan on box2 drops the call into a meetme, creating the room name from the last 2 digits of the current call count -- distributing the calls into 100 meetmes. When I run a script to create 500 call files on box1, box2 starts complaining at 312 calls, logging 'Unable to open DAHDI pseudo channel: Cannot allocate memory' on the console. From the 'callers perspective' the call is dropped between 'There are currently x other participants in the conference' and the 'beep-beep.' 'top' says Asterisk is only using about 1/2 gigabyte of RAM. 'top' says Asterisk is using about 250% of the CPU (4 physical, 8 logical cores). 'ulimit' (added to /usr/sbin/safe_asterisk in the run_asterisk() function) says the open file limit is 397,006. 'ls -l /proc/$(cat /var/run/asterisk/asterisk.pid)/fd | wc -l' says Asterisk only has 2,194 files open. 'iftop' sees about 24Mb of bandwidth in each direction between the boxes. Using confbridge() I can easily get 3,000 calls (14,869 open files, 180Mb bandwidth), but I'd lose some functionality and have to re-write parts of my application. Any clues of what limit I'm hitting and how to increase it? DAHDI has a pseudo channel limit of 512, somebody has already posted how to change it with modprode. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 is more efficient for 1 to many broadcasting?
On Tue, Mar 18, 2014 at 1:02 PM, James Sharp ja...@fivecats.org wrote: Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is MOH is more efficient, can someone suggest a way for a channel to be the source of a MOH class? What sort of channel count are you looking for? We did some load testing recently and found less people in a bridge is better then more. Audio source location didn't really matter much. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Availability with Asterisk
On Thu, Mar 6, 2014 at 10:57 AM, Mitul Limbani mi...@enterux.in wrote: Hello, Using Single Server with multiple VMs essentially kills the purpose, coz it doesnt protect against physical hardware failures. To save costs, use low end box as failover, to keep u in business, till primary box goes live. Correct, in this case para-virt is not the way to go. You'll want to use a virtualization platform that does support multi-hardware with live migration support. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Availability with Asterisk
On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote: Hi Thorolf, Am 06.03.2014 16:21, schrieb Thorolf Godawa: Using (para-)virtualization with Xen could be an other option, on systems with low load this works reliable, but what happens on systems with high load? Are there any issues known about problems with the realtime, packet loss etc. because it runs in a VM? hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this related to high availability? I think it's not. :) I think the way to go for high availability (and scalability) is Kamailio! In a redundant setup, running on 2 separate physical machines (maybe in a VM, doesn't matter). Then you make them failsafe using whatever tool(s) available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio and any of them could fail (but 1 :-) ) and you will still be online. If you want to further develop the high availability thought, then you could use CephFS which will give you self-healing, 100% available storage over multiple physical storage servers. There you could store your Asterisk config files, or your MySQL database used by all the Asterisk servers, for CDRs, SIP registrations etc. It's kinda slow, but I think fast enough for Asterisk / MySQL. :) And, to scale and to make the Asterisk nodes redundant (redundancy is not really needed anymore, since Kamailio takes care of that, but basically then you get also VM/physical redundancy), you could look into OpenNebula which provides a nice auto-scaling feature already out of the box. If there's load on your Asterisk VMs, OpenNebula will detect this and spawn new Asterisk VMs (probably on different physical servers, otherwise it doesn't make that much sense performance-wise) which will automagically receive requests/calls from Kamailio. If the load goes down, the VM can be automagically stopped again to free resources for other VMs/applications. OpenNebula is less popular than OpenStack, which seems to be the first choice for Cloud-stuff today, but what I liked about OpenNebula is that it provides the auto-scaling feature already in the customer-facing web-frontend out-of-the-box, unlike OpenStack. So you could offer your customers a self-managed, redundant Asterisk cloud or something like that. :) In theory, this combination should give you a 100% redundant, auto-healing, auto-scaling VoIP setup. :) +1 to this post. A lot of good information here. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] G729 Licensing Revisited - I'm Sorry!
On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore dar...@moores.ca wrote: On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com wrote: No such thing as 'free open source g729 license', if you actually read the site: There is regarding the copyright on the code. The fact it is also patent encumbered is a different issue. DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. So, basically you are illegal using them if you didn't pay for them. Not true. He said it was a lab setup. It is totally legit to use patented processes in an evaluation/lab environment. Correct, I didn't mention this, since I was assuming OP was talking about getting it into production. Should have been more clear. 3) Is there a performance/stability/security gain when using the commercial vs. open source version or vice versa. See above about about open source license. Your comment about open source is irrelevant to performance, stability, and security. WRT these criteria, I would be surprised if there is much of a difference. The free software isn't locked to a mother board, so that might count towards performance by some measures. Now having said that. I agree once you leave the lab environment and decide you need g.729, you will unfortunatly need a licence to keep using it. The real question is: is there really any choice other than Digium for the licence? Due to the dual licensing of the asterisk code, even if you could license the codec elsewhere, you might be violating Digium's OSS license when you don't but their commercial asterisk license. Cheers, Darryl -- _ -- 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 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] G729 Licensing Revisited - I'm Sorry!
On Thu, Feb 27, 2014 at 5:24 PM, Jayson Devor jayson.de...@gmail.com wrote: Hello Everyone, We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here: http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/ No such thing as 'free open source g729 license', if you actually read the site: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. So, basically you are illegal using them if you didn't pay for them. A little more about our setup. All recordings have been converted to G729, all voicemail messages are also in G729, finally allow=g729, disallow=all is specified in our config. Questions: 1) Is there anything we overlooked in our attempt to implement g729 passthough, and stop all transcoding efforts? 2) do we still need to purchase 23 G729 licenses? If so, is asterisk 10$ license recognized by the patent holders (ie, is Digium authorized to sell the license on behalf of the patent holders)? Yes, getting a license from digium should be sufficient to cover your usage. Plus you'll be supporting the project. 3) Is there a performance/stability/security gain when using the commercial vs. open source version or vice versa. See above about about open source license. I was reluctant to bring this topic up yet again , and yes I did google around and read the different material on the subject however, I am still in need of some definitive answers. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 603 Declined error message
?Set(CALLERID(name)=)) in new stack -- Executing [cidlookup_return@cidlookup:2] Return(SIP/trunk503in-010b, ) in new stack -- Executing [51104@from-trunk:4] ExecIf(SIP/trunk503in-010b, 0 ?Set(CALLERID(name)=3145152244)) in new stack -- Executing [51104@from-trunk:5] Set(SIP/trunk503in-010b, __CALLINGPRES_SV=allowed_not_screened) in new stack -- Executing [51104@from-trunk:6] Set(SIP/trunk503in-010b, CALLERPRES()=allowed_not_screened) in new stack -- Executing [51104@from-trunk:7] Goto(SIP/trunk503in-010b, app-blackhole,hangup,1) in new stack -- Goto (app-blackhole,hangup,1) -- Executing [hangup@app-blackhole:1] NoOp(SIP/trunk503in-010b, Blackhole Dest: Hangup) in new stack -- Executing [hangup@app-blackhole:2] Hangup(SIP/trunk503in-010b, ) in new stack == Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'SIP/trunk503in-010b' Scheduling destruction of SIP dialog '8066eb6f589ce3125b652973b4b00' in 32000 ms (Method: INVITE) --- Reliably Transmitting (NAT) to 172.17.184.46:31285 --- SIP/2.0 603 Declined Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 From: Haley, Scott sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00 To: sip:51...@edj.devjones.com;tag=as06e2e068 Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Server: FPBX-2.8.1(1.8.13.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- SIP read from TCP:172.17.184.46:31285 --- ACK sip:51...@edj.devjones.com SIP/2.0 From: Haley, Scott sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00 To: sip:51...@edj.devjones.com;tag=as06e2e068 Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 ACK Max-Forwards: 70 Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 User-Agent: Avaya CM/R016x.02.0.823.0 Route: sip:192.168.122.51;transport=tcp;lr;phase=terminating Content-Length: 0 - --- (10 headers 0 lines) --- You'll want to talk to the FreePBX guys, as you are just hanging up the outbound call. -- Executing [51104@from-trunk:7] Goto(SIP/trunk503in-010b, app-blackhole,hangup,1) in new stack -- Goto (app-blackhole,hangup,1) -- Executing [hangup@app-blackhole:1] NoOp(SIP/trunk503in-010b, Blackhole Dest: Hangup) in new stack -- Executing [hangup@app-blackhole:2] Hangup(SIP/trunk503in-010b, ) in new stack -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] G729 - what happens if licences used up?
On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield t...@softins.co.uk wrote: I haven't been able to find the answer online, and am not currently able to conduct an experiment to find the answer... I understand that in a SIP call where G729 has been negotiated as the preferred codec, a G.729 licence is not consumed until there is a need to perform transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, etc. What happens when a SIP call in progress needs a G.729 licence and they are all in use already? Does the call fail, or go silent, or do a re-INVITE to negotiate another codec? I'm interested in what happens on Asterisk 1.2 (for a legacy system), and also whether it is any different on later versions. The question depends if you are offering up other codecs or not. If you only using g729, the call will fail to establish because lack of codecs. If you offer a both g729 and ulaw, then ulaw will be used. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 gateway address
On Fri, Feb 14, 2014 at 5:40 PM, Dave Swangler ctit...@live.com wrote: Hello, I inherited an Asterix phone system. I am well versed in Windows based platforms but have zero experience in Linux and Asterix, no make matters worse I have no documentation on this system. I had to change the entire networks gateway address for various reasons but now the Asterix system will not send messages via email. I think it is because of the gateway change. How do I change the gateway address? Is this product something I could contract out to have remote support? Thanks, What you describe is more of a Linux support issue then specific to Asterisk. Depending on your OS, will dictate how to change your gateway. check /etc/network/inferfaces if you are ubuntu / debian. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 does extensions.lua compares to extensions.conf ?
On Thu, Feb 13, 2014 at 1:04 AM, George Joseph george.jos...@fairview5.com wrote: On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote: Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? The lack of replies should give you your answer. Extensions AEL and LUA don't get much action these days, I'm sure there are a few people that use them but extensions.conf has way more code coverage from a testing POV. Your better off using AGI if you want to leverage redis or memcached. Actually, I use Lua dialplans in several production systems. Some are used in conjunction with traditional dialplans and some are the only source of dialplans. They've always been rock solid. I actually find it easier to configure even a moderately complex dialplan than the traditional dialplan syntax. Cool, you are in the minor on that one. My only caution about using them about be the lack of support if you had issues. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 does extensions.lua compares to extensions.conf ?
On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote: Hello, How does extensions.lua compares to extensions.conf or extensions.ael on stability, performance and features ? Would you recommand extensions.lua as an easy/easier way to access memcached, redis or equivalent ? Thoughs ? Comments ? The lack of replies should give you your answer. Extensions AEL and LUA don't get much action these days, I'm sure there are a few people that use them but extensions.conf has way more code coverage from a testing POV. Your better off using AGI if you want to leverage redis or memcached. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 as a media gateway
On Fri, Jan 31, 2014 at 11:27 AM, richard.seg...@marisec.ca wrote: I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to another system (another asterisk box?) for processing. At the end of the day, what I am hoping for is to have 1 brain, and mutiple work horse audio gateways that can be added and removed as needed. Has this been done? Can anyone point me to some documentation on how others have done this? It's always fun to play Yes, this is basically functionality for Asterisk. If you are using SIP, you want to REINVITE media away from your core Asterisk box. I suggest picking up the book[1] and reading the chapter on connecting multiple Asterisk boxes together. [1] http://www.asteriskdocs.org/ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 not receiving call from VPN address
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. You might think Kamailio is transmitting it to Asterisk, however without looking at the actually routing tables on Kamailio you'll never know if it actually made it to Asterisk. Again, we need a pcap trace on both Kamailio and Asterisk, plus what your routes look like (route -n), for a call. It will show us clearly what is happening. This all sounds like a routing issue, so your network admins should be able to help troubleshoot. -- _ -- 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 not receiving call from VPN address
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, Using ngrep/tcpdump shows the packet clearly going from the Kamailio server and arriving at the Asterisk server. This is why it's a mystery that Asterisk doesn't see the call coming in. We tried removing the firewall (so iptables -L shows no rules at all) but that didn't help unfortunately. You might think Kamailio is transmitting it to Asterisk, however without looking at the actually routing tables on Kamailio you'll never know if it actually made it to Asterisk. Again, we need a pcap trace on both Kamailio and Asterisk, plus what your routes look like (route -n), for a call. It will show us clearly what is happening. This all sounds like a routing issue, so your network admins should be able to help troubleshoot. I finally re-read the complete thread. When are you starting the VPN on your Asterisk server, before or after Asterisk has started? If after, and you are binding to 0.0.0.0, it is likely Asterisk is not actually bound to your tun0 interface. So, for a test, explicitly have asterisk listen only on the tun0 interface, retry your call. Or setup your tunnel, then stop Asterisk and start it again, that should cause it to bind properly. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 not receiving call from VPN address
On Tue, Jan 21, 2014 at 5:18 PM, David Cunningham dcunning...@voisonics.com wrote: On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote: (Please don't top-post.) On Wed, 22 Jan 2014, David Cunningham wrote: We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the sip set debug on didn't show the packets. Have you confirmed via 'netstat' (or some other system level toop) that Asterisk is actually listening to UDP port 5060 on the VPN IP address? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Steve, When we have bindport = 172.x.x.14 then netstat -udpln shows the following. When bindport is 0.0.0.0 then netstat shows it listening on 0.0.0.0 as you'd expect. udp0 0 172.x.x.14:50600.0.0.0:* 18114/asterisk -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 At this point in time, you'll need to show us a .pcap on the Asterisk box, when you make a call to it via Kamailio. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Fax detection *11.7
On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger ja...@j-mb.de wrote:
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---
Current Sessions : 0
Reserved Sessions: 0
Transmit Attempts: 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels: 1
Max Concurrent : 0
Success : 0
Switched to T.38 : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 0
Protocol Error : 0
IO Partial : 0
IO Fail : 0
Digium T.38
Licensed Channels: 1
Max Concurrent : 1
Success : 0
Canceled : 0
No FAX : 0
Partial : 0
Negotiation Failed : 0
Train Failure: 1
Protocol Error : 0
IO Partial : 0
IO Fail : 0
so that should be ok.
The corresponding dialplan section starts with
[from-sip]
include = inbound
[inbound]
exten = _X.,1,Answer()
exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
exten = _X.,n,Ringing
exten = _X.,n,Progress()
exten = _X.,n,Wait(5)
exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
...
exten = fax,1,NoOp( FAX DETECTED )
exten = fax,n,Goto(fax-rx,receive,1)
in the sip.conf i specified
[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
qualify=yes
nat=force_rport,comedia
faxdetect=yes
t38pt_udptl=yes
...
[abcde]
type=peer
insecure=invite
defaultuser=12345678912
fromuser=12345678912
fromdomain=abcde.ab
secret=guess-what
host=abcde.ab
qualify=yes
context=from-sip
dtmfmode=rfc2833
callbackextension=12345678912
but all i can see if i try to send a testfax is
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, )
in new stack
0x7fd11404cd00 -- Probation passed - setting RTP source address to
123.456.789.123:17108
-- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
0?black,1) in new stack
-- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
in new stack
-- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
new stack
-- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
SIP/123SIP/456,30,oxX) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- Called SIP/200
-- Called SIP/201
-- SIP/123-0018 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/456-0017 connected line has changed. Saving it until answer
for SIP/abcde-0016
-- SIP/123-0018 is ringing
-- SIP/456-0017 is ringing
Don't expect T.30 over SIP to be reliable. If you need fax, you should
be using T.38. Your codec is likely the issue.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk not receiving call from VPN address
On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham dcunning...@voisonics.com wrote: Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers. Thanks in advance for any help. The ngrep on the Asterisk server: U 2014/01/17 13:15:15.599557 172.x.x.x:5060 - 103.y.y.y:5060 INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0. Record-Route: sip:172.x.x.x;lr=on. Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0. Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997. From: 9067271 sip:9067271@172.x.x.x;tag=198791249. To: sip:9067268@172.x.x.x. Call-ID: 1905625787@192.z.z.z. ... 172.x.x.x is the Kamailio server's VPN address 103.y.y.y is the Asterisk server's real address 192.z.z.z is the calling phone's LAN address Sounds like a routing problem opposed to an application issue. You'll have to fire up tcpdump on Kamailio and see what happens to the packet. The look at the local routing tables to see where it is getting routed. If Asterisk is not receiving the patch, then Kamailio is not routing it properly. You'll be able to see everything once you have a pcap of the call. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 not receiving call from VPN address
On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? Well, you need to use tcpdump on each hop across your network. If are Asterisk is not getting anything, either it is not receiving anything (check transmit side) or the firewall is dropping it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 install TEST_FRAMEWORK(E) ?
On 14-01-17 12:55 PM, Richard Mudgett wrote: On Fri, Jan 17, 2014 at 3:42 AM, Olivier oza.4...@gmail.com wrote: Hi, I've installed a brand new Asterisk 12.0.0 system in which I can see, with make menuselect, in Test Modules tab, that each test entry such as test_acl can't be installed due a to missing TEST_FRAMEWORK(E) dependency. Where this TEST_FRAMEWORK(E) comes from ? How can it be installed ? I've installed testsuite according https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suitebut stopped before installing Third party modules (lua-devel and on). TEST_FRAMEWORK is an option selectable under the Compiler Flags - Development menu in menuselect. ./configure --enable-dev-mode -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 11 and H.323 trunk using OOH323 - is it stable?
On 14-01-16 03:37 PM, Gergely Kiss wrote: Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as plan B). As I never worked with H.323 channels in Asterisk earlier, I'm not sure if it's stable enough to be used in production. Googling about the subject didn't help much, I could only find some old and probably outdated information which I don't want to rely on. Can you please confirm if the OOH323 module in Asterisk 11 is stable enough to use for voice calls? No extra functionality is needed, just to be able to create a H.323 trunk towards the provider and make and receive a maximum of 30 simultaneous voice calls through the trunk. Thanks for your kind response! Save yourself time / energy and insist using SIP. If your ITSP cannot accommodate your request, thank them and look for another provider. H323 is Asterisk is basically dead, sure there is a module, sure it might compile, but you'll be going down the path of zero help. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Does cdr adaptive odbc re-connect automatically after a long idle time?
On Sat, Jan 11, 2014 at 4:56 PM, Charles Wang lazy.char...@gmail.com wrote: Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR messages in the CLI at this period. Could you please tell me which settings are wrong? Why dose my odbc connection not re-connect to MySQL automatically? I checked the setting below: CLI: ubuntu*CLI cdr show status Call Detail Record (CDR) settings -- Logging:Enabled Mode: Simple Log unanswered calls: Yes Log congestion: Yes * Registered Backends --- cdr-custom Adaptive ODBC csv ubuntu*CLI odbc show all ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 2014-01-11 18:16:40 Pooled: Yes Limit: 1000 Connections in use: 0 -- /etc/asterisk/cdr.conf lists below: [general] enable=yes unanswered = yes congestion = yes endbeforehexten=yes [csv] usegmtime=no; log date/time in GMT. Default is no loguniqueid=yes ; log uniqueid. Default is no loguserfield=yes ; log user field. Default is no accountlogs=yes ; create separate log file for each account code. Default is yes -- /etc/odbc.ini [asterisk-connector] Description = MySQL connection to 'asterisk' database Driver= MySQL Database = mydatabase Server= localhost UserName = root Password = mypassword Port = 3306 Socket= /var/run/mysqld/mysqld.sock -- /etc/asterisk/res_odbc.conf lists below: [ENV] [asterisk] enabled = yes dsn = asterisk-connector password = mypassword pre-connect = yes sanitysql = select 1 pooling = yes idlecheck = 30 share_connections = yes limit = 1000 connect_timeout = 60 negative_connection_cache = 600 -- /etc/asterisk/cdr_adaptive_odbc.conf lists below: [cdr] connection=asterisk table=cdr alias start = calldate alias phoneno = phoneno alias userid = userid alias callerid = callerid I would be inclined to check the database side over asterisk. We use almost the same setup and don't have any issues. We go some time 12 hours between calls. Once thing you could do is enable debug logs and see what Asterisk is doing when the odbc connection is down. EG: it should be attempting to reconnect. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible? Well, if you want anybody to call you, you need to leave it open to the public. Meaning, you can't really secure it. Obviously, don't have any outbound trunks configured on the box so that the only location some could dial would be your extension. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 read IRQs and timing slips values
into the software. All that to say, try upgrading DAHDI and see what happens. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Dropped call on new CISCO router for no reason!
On 14-01-06 09:27 AM, Nick Cameo wrote: Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly appreciated, Nick. Show us the problem, give us a SIP trace[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Question about Management/Control Protocol Licensing
On 13-12-11 03:15 PM, Steve Murphy wrote: I see the following paragraph in the Asterisk trunk LICENSE file: In addition, Asterisk implements two management/control protocols: the Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface (AGI). It is our belief that applications using these protocols to manage or control an Asterisk instance do not have to be licensed under the GPL or a compatible license, as we believe these protocols do not create a 'derivative work' as referred to in the GPL. However, should any court or other judiciary body find that these protocols do fall under the terms of the GPL, then we hereby grant you a license to use these protocols in combination with Asterisk in external applications licensed under any license you wish. This probably originated some years ago, and I wonder if Digium or the Asterisk community might consider adding the OTHER management/control protocols to this list: ARI, and the ExternalIVR interface. If not, it might be instructive to learn why! Would also like to see this update to include ARI. We talked a little about it at astridevcon, and I think it is likely an oversight. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Call Queue advise
On 13-12-09 06:47 PM, Bryan Anderson wrote: I have a call queue that rings about 15 users and they are wanting to set it up so that the last person to answer a call doesn't ring on the next incoming call. What would be the best way to handle this? I have been looking at the strategies and none of those seem to be right for this. My current thoughts are probably a macro that places a penalty on the user tell the next call is answered. Any advice for this would be greatly appreciated. You have agents that log into a queue that don't want to get calls? Is that what you are saying? Options 1 - log the agent out, they don't get the next call. Option 2 - Set up weights for your agents, as answer a new call, increment then up so they don't get the next. Either way, I see issues with the setup. Best ways is to rethink your queue strategy and stop using ring all. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Windows
On 13-12-04 10:19 AM, CDR wrote: Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. Note: I am a Linux expert user, but it took me years to get here. And still, moving from regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET framework and Windows server 2012 are miles away in terms of friendliness and on equal footing on performance. I don´t mean another slow cygwin port, I man a native Asterisk for windows. In fact, I would invest on the project if somebody wants to do it. Do you just sit around and think shit up to blame Digium all day? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] issue with speech in IVR
On 13-11-28 05:22 AM, Salaheddine Elharit wrote: hello, i have add the the code below but the issue still the same i can't go to the project during the speech any other solution best regards NB:for the version of asterisk i can't move to another version for the moment exten = _X,1,NoOp(Digit entered during prompt) exten = _X,2,Goto(project,s,1) Then you have a DTMF issue, Background will allow DTMF to interrupt the prompts. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 uses 105% CPU
] features.c ast_features_init() 0x7f993c0fe700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f993c17a700 do_devstate_changes started at [ 750] devicestate.c ast_device_state_engine_init() 0x7f993f17b700 logger_threadstarted at [ 1143] logger.c init_logger() 0x7f993f1f7700 listener started at [ 1483] asterisk.c ast_makesocket() 0x7f993f273700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 55 threads listed. First thing, prune your Asterisk configuration and don't load any modules you don't need to use. Are you really using chan_mgcp, chan_skinny, res_calender, etc. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] issue with speech in IVR
On 13-11-27 04:57 PM, Salaheddine Elharit wrote:
hello list
i have an IVR menu in asterisk 1.4
like below
exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)
[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}music1)
exten = s,n,Background(${sounds_path}music2)
exten = s,n,Background(${sounds_path}music3)
exten = s,n,WaitExten(5)
exten = s,n,goto(home,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,WaitExten(5)
exten = i,n,goto(home,s,1)
exten = 1,1,Goto(project,s,1)
exten = _X,1,NoOp(Digit entered during prompt)
exten = _X,2,Goto(project,s,1)
[project]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}mymusic)
exten = s,n,WaitExten(5)
exten = s,n,Goto(project,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,goto(project,s,1)
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
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Re: [asterisk-users] Asterisk uses 105% CPU
On 13-11-27 07:35 AM, Jonas Kellens wrote: Server specs : XEON E3-1220V2 4 GB RAM 2 x 500GB HD (RAID0) 1 U HOT-PLUG PSU Linux sip.server.tld 2.6.32-358.18.1.el6.x86_64 #1 SMP Wed Aug 28 17:19:38 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux There is no transcoding. Calls are using G711a. Maybe there is some trancoding when using voicemail... How can I find out if there is trancoding ?? Maybe explain what your dialplan is doing. Are you making system calls to a database or AGI? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] CEL for attented transfer
On 13-11-19 11:03 AM, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jairo, Le 19/11/2013 01:36, Jairo a écrit : https://wiki.asterisk.org/wiki/display/AST/CEL+Events+and+Fields Thanks for your reply, but I have read this page of the wiki, I know what the fields mean. Well, it is a way lot harder to figure out because you used features.conf. Because of this, local channels are involved. What I don't understand is how the events in my example can be used to determine 107 was attended transferred to 103 by 100. Or I do know that Local/103@100-0042;1 and Local/103@100-0042;2 were created by asterisk when SIP/100-0275 asked for atxfer? How does the event ATTENDEDTRANSFER/ SIP/107-0274/ Local/103@100-0042;1 show that 107 is transferred to 103? Specifically, you are going to have to track the channel IDs and look at the sequence of events. Then make an educated guess about what is happening. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] calendar.conf include
On 13-11-13 10:20 AM, Jonas Kellens wrote: Hello, can I use include-statements in the calendar.conf configuration file ? You _should_ be able to use it will every .conf file, otherwise it is a bug. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Unix connections not always disconnecting
On 13-11-07 10:31 AM, Ishfaq Malik wrote: On 7 November 2013 15:26, Gareth Blades mailinglist+aster...@dns99.co.ukwrote: On 07/11/13 11:20, Ishfaq Malik wrote: Hi We are using asterisk 1.8.23.1 We have a script that runs on a minute cron which polls the asterisk server for 3 bits of information by using asterisk -rx 'command' which then gets pushed to a graphite server we have 99% of this runs smoothly. Out of interest what are you trying to monitor? We tend to use cacti for graphing and snmp provides all the information we require. Active calls, sip peers connected, sip peers disconnected and then breaking all of those down by customer as we run a multi tenanted set up. SNMP would give us totals but I don't think it would do the breakdown by customer. You should avoid using the CLI to access that information. You'd likely getter better results using AMI or CEL. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Capture dead phone?
On 13-11-07 07:51 PM, Mitch Claborn wrote: Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will knock something loose with their foot. If/when that happens while the agent is on a call with a customer, I'd like to be able to save that caller and put them back in the queue (at the head of the queue). No, you won't be able to save the call if the far end goes down. Best you could do would be to enable qualify, track then the agent phone goes offline, if a call also drop around that time frame, initial some sort of callback. However, solve the issue at the source. Spend the money for a UPS at each desktop, convert your phones to PoE and install a UPS in your server room. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Tired of dropouts and garbled phone calls - where to go next?
On 13-10-28 06:03 PM, Patrick Lists wrote: On 10/28/2013 07:29 PM, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine That's a good start. Now what have you done to conclude that the Asterisk server is not the cause of your problems? More than enough bandwidth That's irrelevant. It's about the quality of that bandwidth. Have you figured out if there might be a lot of packetloss or are you perhaps on a cablelink which is a *shared* medium? Once your link hits the box in the street it shares it with others who might be eating up all the bandwidth with their torrent downloads etc.? Use tools like iperf, smoke ping and mtr to see if there are obvious problems on the route to your VoIP provider. Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure what option would be the best Once the packets leave your premises and your ISP/cable company starts messing with them a QoS setting is generally not honored so not very helpful unless your LAN is congested. Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use If those analog lines are cheap, easy to get then as an intermediate solution I would order those analog lines as fast as I could. Or fix the VoIP problems, whichever is faster. Hire a consultant An experienced VoIP consultant should be able to tell you what is or could be causing your problems. With your users sick of phone service it suprises me that you haven't already hired one. Ditch the system and buy a pre-packaged system - RingCentral or some such. And what if it's your Internet link or the route to your VoIP provider? What if your VoIP provider is messing up? There are no local asterisk professionals who can help, and we are a little leery of opening up our system to outside consultants. If you don't want that then you don't want that but given the state your users are in I would be less worried about giving a Consultant access to the Asterisk box and more worried about my job :-) Anyone else face the above, and finally abandoned Asterisk for a commercial system? I have seen that once years ago where some clueless sales guy had totally oversold an ancient Asterisk/Bristuff/ISDN setup which was very buggy and crash prone. There was no way to make that work reliably. After the supplier failed for months I was brought in to review the setup and possibly fix it. Told the customer to cut its losses. So they kicked out their supplier and opted for a different setup. We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. I don't know how Grandstream is these days. I thought the GXP2100 was ok but I guess you already know if there's a problem with those phones from the (lack of) intra-office call complaints from your users. Suggestions welcome. Hire a Consultant or someone who has been part of this Community for a while and is well known on this list or in #asterisk on irc. Provide remote access if required. Change passwords afterwards. If you really don't want to provide remote access then find a reputable VoIP provider with a switch physically as close as possible to your location, get a DID for a few bucks, hook it up to your Asterisk box and route it to a line on your phone, grab your cell, call that DID and see if you still have the problem. It wouldn't be the first time that the link between you and your VoIP provider just doesn't cut it. Or maybe your VoIP provider just sucks and you need to change to a different one. Both flowroute.com and voip.ms work well for me (no affiliation). Or maybe your Internet link sucks and you need to change your ISP. ^ this Like others said, you really need to drill down and find out where your audio issues are. Local is easy to do, since you control the network, remote is harder. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Is this big of new modification in Asterisk Events Objects values ?
On 13-10-25 07:06 AM, virendra bhati wrote: Hi Team, Thanks for your great job an Asterisk new features developments. I installed asterisk-12 Beta and found some changes as well which i notice to put in-front of your knowledge, don't know that bug of new modification into objects or old version (asterisk-11) mistake corrected that time, anyway *Asterisk-12:* Array ( [Event] = ConfbridgeMute [Privilege] = call,all [Conference] = 42 [BridgeUniqueid] = 9f2ae5df-0749-4494-b8b7-12eb50dc765d [BridgeType] = base [BridgeTechnology] = softmix [BridgeNumChannels] = 2 [Channel] = SIP/5000-0006 [ChannelState] = 6 [ChannelStateDesc] = Up *[CallerIDNum] = 5000* [CallerIDName] = 5000 [ConnectedLineNum] = unknown [ConnectedLineName] = unknown [AccountCode] = [Context] = from-sip [Exten] = 1234 [Priority] = 3 [Uniqueid] = 1382599433.22 ) Please check the BOLD section. earlier is was *[CallerIDnum] * *So 'n' is now 'N' * Asterisk AMI got basically a rewrite[1] of how it works, so there are some breaking changes moving forward. Read ChangeLog and UPGRADE.txt in the source tree for more information. [1] https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Calls Recording Solution
On 13-10-21 10:39 PM, bilal ghayyad wrote: Hello; I am looking for calls recording solution to do recording based on the network traffic .. The solution to be competitive and appreciate if it is open source .. Any suggested one? http://www.orecx.com/ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 events is generated as Asterisk Manager logoff
On 13-10-14 12:31 PM, virendra bhati wrote: As I said, I am running a event capture program and it looks for Events and work on the basis of events. But some time it stop working so I want to auto-connect with asterisk back as it was disconnect with asterisk AMI. Well, if the socket is closed, you lost the connection. Open the connection again and profit. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Capture Media IP in CDR (CDR)
On 13-10-13 03:06 PM, CDR wrote: I am quite surprised about the degree of surprise in the group. A few days ago, somebody called a school and issued a threat, through my network. The call came from China, but of course it was US caller. The DA wants to know where call came from. The caller ID is Restricted and the chinese carrier is playing games. If I had a way to store the media IP, I would be able to pinpoint the offender in the US, or the company that touched the media last. As a result of Asterisk not having this functionality, many children are danger and this country at large is at a great peril, since Asterisk is the most widely used low-cost technology for telecommunications. I need Digium to store this IP in the CDR. I will be honest with the government and let them know that my tool is incapable of saving lives or safeguarding our national security because nobody thought about this. PD: I am not paying for a patch, since this is huge burden on a small company like mine, with a single employee, and also because the whole world will enjoy the benefit. It is not fair that I would have to hire somebody to patch Asterisk. I appeal to Digium to patch Asterisk. Don't worry about it, I'll step up and pay for the patch. No need for you to waste your profits on something this. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 cpu average load
On 13-09-05 03:11 PM, Steve Edwards wrote: On Thu, 5 Sep 2013, Kamlesh Kumar wrote: Running one asterisk server with below details. Only SIP to SIP calls. No real time configuration, no recording, no voicemail, no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150 to 200 concurrent calls and we start getting problem in call quality like delay in connectivity, voice breakage etc Hardware: 2 Physical processor Intel(R) Xeon(R) CPU5120 @ 1.86GHz 8 GB RAM 500 GB Sata HDD Asterisk: 1.6.2.9 PHP 5.3.3 (cli) MySQL: 5.0.77 Linux: CnetOS 5.5 (Final) Please suggest the solution. Need a bit more detail. The 5120 is kind of a wimpy processor, but what is keeping it busy? What do 'top' and 'htop' show are consuming the processor? What is your application? What are 200 calls doing? Are you calling a bunch of AGIs written in scripting languages? Eliminating translation is difficult. How do you know you were successful? Do 'module show like codec_' and 'module show like format_' show anything unexpected? Also drop Apache and Database from your PBX. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 member ackcall - cpuspikes
On 13-08-07 08:42 PM, zendel fernandez wrote: hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall Scenario: 1. User calls in a General Number 2. Call is queued in Queue Application 3. Queue calls a Local/@members channel 4. At members context: Dial The real member(called party) channel with a U(GOSUB X) routine 4.1 The called party answers, is led to the GOSUB routine X: Here the prompt is given to the called party to acknowledge the incoming call [ depending on the out put, this will return appropriate GOSUB result ] 4.2 Based on the GOSUB result, the Dial proceeds 5. The Queue proceeds based on the result taken at 4.2 above. i.e. Take it as a success build the bridge between the caller member Whether to DIAL the next member The Question: All goes well the dial-plan works. If between step 4.1 4.2, the caller hangs up asterisk gives CPU spikes. Symptom: ASTERISK CLI gets stuck until step 4.2 returns. Console Error: app_dial.c: Could not stop autoservice on calling channel [ Somehow get the feeling that this is not the real error] What could be the reason for CPU SPIKES. How to avoid this ? What are you doing in your GOSUB X routine, you are likely blocking the thread in Asterisk, which is causing your autoservice errors (and yes, they are real errors) which increases the CPU on asterisk. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 CPU use
On 13-07-29 10:22 AM, Eduardo Leones wrote: Hello, working in a call center where we set up a structure in asterisk. When my voip reaches 150 calls are with bad quality. We do not transcode codec. What I realized using the top command server (CentOS) processing is too high for the asterisk. But the general processor server is down. Would any limitation of Asterisk to use more hardware resources? Your load average is insane. Time to off load resources from your PBX, for example why are you running httpd? You need to figure out where your bottleneck is and then adjust it. Using something like iotop, netstat and see what your system is doing. I doubt this is a CPU issue. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Fw: Stress testing Asterisk
On 13-05-22 10:02 AM, Tommy Cooper wrote: From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system. - Forwarded Message - From: Mitul Limbani mi...@enterux.in To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 22, 2013 3:23 PM Subject: Re: [asterisk-users] Stress testing Asterisk I have a question here. How can we test the quality of voice upon increasing the call load? Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario? Once upon a time, we set out to create exactly this for testing asterisk. Our goal would have been to run the test every week, comparing the results from the previous week, to make sure asterisk's performance was not getting worse as new commits happened. We came up with the idea of loading testing asterisk using SIPp or some other dialer, then determining at what point asterisk would start failing (performance). We decided the point of failure was quality of audio, since it is usually the first thing to go (even though call control still works). It took a while, but with the help of Leif, we found a tool to analyse audio streams (using MOS score[1]). Basically, you take the original audio file, play it across the network, then record the other side. Then, comparing the two files via Aqua, you get your MOS score. If the score was less then x, you knew asterisk was hitting a performance limit. Track that over time and concurrent calls, you have your metrics. [1] http://www.sevana.fi/aqua.php -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 12 and OPUS Codec
On 13-05-10 02:45 PM, James Mortensen wrote: I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS codec, which is part of the WebRTC standard as the default codec. Doubt it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] ACD problem
On 13-04-10 04:08 PM, Tommy Cooper wrote: Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want to design a system where customers can call my number, that call will then be directed to either extension 1000 or 1001. If both extensions are in use, I want that 3rd call to be queued. I don't think that the config below will direct calls to extension 1001 because the second line states that any incomming calls should be routed to extension 1000. How do I change this so that calls are directed to all of my exensions? Forget dialling the phones directly, let the queue deal with it. Dump everything in to the queue, then just wait for somebody to answer. extensions.conf [from-myprovider] exten = *DID number*,1,Answer exten = *DID number*,2,Dial(SIP/1000) exten = *DID number*,3,Queue(support) ;not sure if this line belongs here exten = *DID number*,4,Hangup queues.conf [general] [support] musicclass=default strategy=rrmemory joinempty=no leavewhenempty=yes ringinuse=no Member = SIP/1000 Member = SIP/1001 agent = 1000,1000 agent = 1001,1001 When using the current config the caller will listen to the 'music on hold' until the agent answers but calls are only being forwarded to extension 1000 as stated above -- _ -- 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 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Peaking and 91 Calls And not a Dime More!
On 13-04-09 02:49 PM, Nick Khamis wrote: Hello Everyone, We are running some torcher tests on our * box using SIPP. The overall idea of the test is to contact asterisk and play a g729 encoded recording. On the asterisk side, we are initiating the echo app for the contacted extension, simulating a two way conversation. For some reason we cannot get past *91* calls on every test, with a lot of resources left: *top* top - 14:28:45 up 1 day, 1:45, 2 users, load average: 1.09, 0.80, 0.59 Tasks: 56 total, 1 running, 55 sleeping, 0 stopped, 0 zombie %Cpu(s): 7.6 us, 8.5 sy, 0.0 ni, 82.7 id, 0.0 wa, 0.0 hi, 1.2 si, 0.0 st KiB Mem: 3825108 total, 164480 used, 3660628 free,16324 buffers KiB Swap: 2097148 total,0 used, 2097148 free,97404 cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 7229 root 20 0 70400 25m 5808 S 35.5 0.7 7:29.26 asterisk *iftop* Press H or ? for help 1.91Mb 3.81Mb5.72Mb 7.63Mb 9.54Mb └┴─┴───┴┴ test.example.com = 192.168.2.100 1.75Mb 1.75Mb 1.71Mb = 1.70Mb 1.70Mb 1.66Mb test.example.com = db.example.com 37.3Kb 37.3Kb 36.5Kb = 10.1Kb 10.1Kb 9.87Kb TX: cumm: 8.28MB peak: 1.79Mb rates: 1.79Mb 1.79Mb 1.74Mb RX: 7.93MB 1.72Mb 1.71Mb 1.71Mb 1.67Mb TOTAL: 16.2MB 3.51Mb 3.50Mb 3.50Mb 3.41Mb The SIPP Results -- Scenario Screen [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 10.0(0 ms)/1.000s 50602089.21 s20802 192.168.2.10:5060(UDP) 0 new calls during 0.000 s period 0 ms scheduler resolution 0 calls (limit 100)Peak was 91 calls, after 9 s 0 Running, 332 Paused, 0 Woken up 0 dead call msg (discarded)0 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg INVITE -- 20802 0 0 100 -- 20802 0 0 0 180 -- 0 0 0 0 200 -- E-RTD1 20802 0 0 0 ACK -- 20802 0 [ NOP ] Pause [ 8000ms] 20802 0 [ NOP ] Pause [ 1000ms] 20802 0 BYE -- 20802 0 0 200 -- 20802 0 0 0 -- Test Terminated - Statistics Screen --- [1-9]: Change Screen -- Start Time | 2013-04-09 14:08:07:797 1365530887.797642 Last Reset Time| 2013-04-09 14:42:57:025 1365532977.025339 Current Time | 2013-04-09 14:42:57:025 1365532977.025537 -+---+-- Counter Name | Periodic value| Cumulative value -+---+-- Elapsed Time | 00:00:00:000 | 00:34:49:227 Call Rate |0.000 cps |9.957 cps -+---+-- Incoming call created |0 |0 OutGoing call created |0 |20802 Total Call created | |20802 Current Call |0 | -+---+-- Successful call|0 |20802 Failed call|0 |0 -+---+-- Response Time 1| 00:00:00:000 | 00:00:00:003 Call Length| 00:00:00:000 | 00:00:09:010 -- Test Terminated Can we clear OS and * bottlenecks down into the different parts: OS - Simple commands such as ulimit etc... Asterisk - Startup directives that will increase whatever (i.e., allocated memory, -p value) before addressing hardware resources? Your help is greatly appreciated, Nick. You failed to say what happens when 92 channels are created. Show us your errors. -- Paul Belanger
Re: [asterisk-users] Feature request: Need to INVITE to peer with other domain without peer domain addition
On 13-04-01 03:16 PM, Dmitriy Serov wrote: 31.03.2013 23:15, Barry Flanagan ?: On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com mailto:serov@gmail.com wrote: Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten = s,n,Dial(SIP/peer1/num...@domain2.com mailto:num...@domain2.com,60,r) [peer1] type=friend host=domain1.com http://domain1.com fromdomain=domain1.com http://domain1.com As a result in SIP packet uri: num...@domain2.com@domain1.com http://domain1.com I need: num...@domain2.com mailto:num...@domain2.com I can't use SIP uri dial, i need authorization (peer1) I think asterisk can't do that. Is where work around? Would it work if you created a sip peer [domain2.com http://domain2.com] and set outboundproxy=domain1.com http://domain1.com then sent the call to SIP/num...@domain2.com mailto:num...@domain2.com ? -Barry does not matter. [skype.ippi.com](srv-options-common) type=friend secret=xxx host=ippi.fr fromdomain=ippi.fr outboundproxy=ippi.fr exten = 22,n,Dial(SIP/lo...@skype.ippi.com,60,rS(1200)) INVITE sip:lo...@ippi.fr SIP/2.0 Via: SIP/2.0/UDP 109.60.163.xx:5060;branch=z9hG4bK60e845b5;rport Max-Forwards: 70 From: demon sip:usern...@ippi.fr;tag=as518b59df To: sip:lo...@ippi.fr and exten = 22,n,Dial(SIP/skype.ippi.com/lo...@skype.ippi.com,60,rS(1200)) do: INVITE sip:lo...@skype.ippi.com@ippi.fr SIP/2.0 I studied the source code and found no ways to implement it :( Dmitriy. How about: exten =22,n,Dial(SIP/skype.ippi.com!lo...@skype.ippi.com,60,rS(1200)) -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 1.8 as text to speech server
On 13-03-13 12:50 PM, Amit Salunkhe wrote: On Mar 13, 2013 10:16 PM, Amit Salunkhe amitsalunkh...@gmail.com wrote: Hi I want to know asterisk 1.8 as text to speech server. If we can use as TTS server then it support SSML. Any sample configuration available for this requirement. Plz help me with support asterisk as tts server. Amit-- Asterisk is not a TTS server, it is a PBX. I'm sure you could hack stuff together, but you'd be better off to use external services for TTS. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 AMI - Create a daemon (background process)
On 13-02-24 07:30 AM, Shahid H wrote: I wanted to create a daemon (background process) in PHP. A daemon will use socket to connect with Asterisk AMI to send events and listen the actions. A daemon will also listen the commands from agents via HTTP, for example: A agent pressed a hang up button on a browser - it will send http command to a daemon. A daemon received a command and will then send Hang Up Action to AMI. How should a daemon process be designed to listen multiple actions and events? For example: 50 agents currently on the calls and how should a daemon to monitor the Actions/Events from 50 agents? You don't want to use PHP for your daemon, change to another scripting language (EG: python). -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Realtime vs Static Files
On 13-01-23 04:41 AM, Dan Journo wrote: Hi, We're trying to decide whether to switch back to a static file for sip.conf. Currently we use mysql realtime but can't see any real benefit. Why would someone choose realtime sip over static files? Thanks I'm interested in the feedback too. For years I've used static files for my needs, and worked around some of the 'realtime' limitation with them with other tools. For what we are doing, it works great. Recently though, I've been more and more considering using res_config_curl, which uses the realtime interface, and replace some of the existing toolsets. Sadly, documentation is weak, and I don't suspect it gets much love in production. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Adding custom HTTP headers to Asterisk
Anybody using Apache to proxy HTTP traffic to Asterisk HTTP? I got a request from a developer to add some CORS headers[1], for an application we are writing, and wanted to see if anybody else has had success. [1] http://enable-cors.org/ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
On 12-12-04 12:59 AM, Earl Ruby wrote: If you are trying to provide CDR files to a billing service, such as WebCDR.com, you need to provide files containing your latest call data every 15 minutes or so. I wrote a script and a cron job that will create a new CDR file every 15 minutes with the latest CDR records, without interrupting call flow. You do not need to make any changes to your Asterisk configuration to use these scripts. There are two files that you need to install on your Asterisk server: asterisk-cdr-rollover.sh – A bash shell script. Copy this file into /usr/local/sbin. This script moves the file /var/log/asterisk/cdr-csv/Master.csv to a new file named /var/log/asterisk/cdr-csv/cdr-MMDDHHMISS.csv, where MMDDHHMISS is the current time. A new zero-byte Master.csv file is created using the default umask of the user running the asterisk process. Asterisk will start writing to the new Master.csv file at the end of the next call. asterisk-cdr-rollover – This is a cron job. Copy it into /etc/cron.d and it will run the /usr/local/sbin/asterisk-cdr-rollover.sh script once every 15 minutes. The cron job is set up to run as the user “asterisk”. If you are running asterisk as “root” or some other user name, edit the asterisk-cdr-rollover cron job and change the name of the user running the script to the same name as the user running the asterisk process. The latest versions of these two files can be downloaded from GitHub: https://github.com/earlruby/asterisk-cdr-rollover. Why not use logroate? $ man logrotate -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
On 12-12-04 10:02 AM, Danny Nicholas wrote: IIRC log rotate only rolls the files in /var/log/asterisk, not /var/log/asterisk/cdr-csv You need to configure logroate with the path and filename. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 roll-over / move / rotate an Asterisk Master.csv call detail record (CDR) file every 15 minutes
On 12-12-04 01:05 PM, Earl Ruby wrote: Paul: Four reasons not to use logrotate: 1. logrotate does not provide log rotation every 15 minutes. Sure it does, you can invoke it using crontab, like you do for your script. 2. logrotate will not create unique file names unless you use a date format in the name (file names with a .nnn extension get reused over time), but since logrotate only supports MMDD, not hours, minutes, or seconds, once again you're limited to daily roll-overs. (asterisk-cdr-rollover generates file names using the format cdr-MMDDHHMISS.csv.) Not so, logroate actually supports strftime %s, so you get the number of seconds since the Epoch. Easily converted into any datetime format you wish. 3. My customers need to be able to feed CDR files into a telecom billing, monitoring, fraud-detecting system I work on called WebCDR.com, which works best if it gets a new CDR file every fifteen minutes. postrotate/endscript should work for this. 4. If there are no calls after 15 minutes, with asterisk-cdr-rollover I get a zero byte file, which can trigger a no calls alarm, alerting me that something is wrong with the Asterisk switch. (The switch setup I'm working will always have some calls within a 15 minute block if everything is working correctly, although people with less-busy switches can always configure a larger alarm window to suit their situations.) If you're running a high-traffic switch, this can be a life-saver. No native support, but you could invoke logic using postrotate/endscript from above. If you want a copy, go to https://github.com/earlruby/asterisk-cdr-rollover and click the ZIP button to download the script and cron job. It's free. Understood. If it works for you that is great, I was mostly trying to understand why you choose to rewrite logrotate :) -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 repository for Ubuntu
On 12-11-17 06:23 PM, Mitch Claborn wrote: Is there an Asterisk repository for Ubuntu that has recent versions (e.g. 11)? The standard Ubuntu repository for Ubuntu 12.04 is stick at 1.8. None that I know of. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Managing complex setups with Asterisk
On 12-11-08 01:41 AM, martin f krafft wrote: also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.07.2340 +0100]: What is your point of pain? Right now we do most of the configuration, provisioning, and system management outside of asterisk. My systems are already managed automatically, thankfully no longer with Puppet. ;) I am only talking about configuration of Asterisk, whether in /etc/asterisk or some sensible external data source. My point of pain is the complexity due to a couple of special cases, e.g. - Roaming users, i.e. no 1:n relation between sites and users; - Multiple devices per user (some want them all to ring, some want individual extensions but shared voicemail, …) - Keeping track of the mappings between incoming calls (from SIP providers) and extensions to ring (using incoming contexts and extension groups for that) - Keeping track of which extension uses which outgoing trunk - … With a logical naming scheme, a policy and include files, this is all working. But it's very error-prone and there is a bit of redundancy in the information, so I was wondering if there wasn't a better way. Either way, don't manually build your 6th machine. Start from fresh using some sort of automated tool (chef / puppet). This will help you get on the right path. The new machine for the 6th site is up and running (provisioning (not image-based) took less than half an hour). What now? ;) Then you are on the right path. Either way, it sounds like you need to store your data some place and start building it out. I don't know of any existing tools to do that, and I'm in the same boat. I have everything I want / need managed by puppet, but more dynamic data needs to be moved out into something else. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Managing complex setups with Asterisk
On 12-11-07 05:41 AM, martin f krafft wrote: Hello, we are finally going to redesign our Asterisk-Setup, which has grown quite complex. We have five sites with a total of 400 users, 15 SIP registrations and 3 IAX registrations. We do not use any VoIP-hardware, so it's all software-based. But we make heavy use of features, including voicemail, followme, conferencing, call-recording, and queuing. As I said, the configuration has grown quite complex — so complex that we are all a bit scared to touch it. It works, but as we are now adding a sixth site and upgrading the hardware, we thought it would be a good opportunity to get the sixth site up and running on a new box, then migrate the other sites. Now we are trying to figure out how to organise sip.conf, iax.conf and extensions.conf. I read about Realtime configuration, but I was a bit disappoointed because it's really just moving the section-key-value store from the flat files to a relational database without really making use of any relational features. Sure, it's realtime thereafter, but not any less complex. So what to do? Does anyone have a similar setup and would like to offer a glance into their configs? Are there best practices? Or is there maybe even software (Linux) to manage setups? What is your point of pain? Right now we do most of the configuration, provisioning, and system management outside of asterisk. So, for example here is how we handle the initial configuration and provisioning, we use puppet[1]. Leif and I did a talk at astricon[2] about it. System management is another issue, but we mostly use puppet or external agi / database to control certain _dynamic_ features. Either way, don't manually build your 6th machine. Start from fresh using some sort of automated tool (chef / puppet). This will help you get on the right path. [1] https://github.com/kickstandproject/astricon-2012-presentation [2] http://goo.gl/T8lJR -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Parameterize asterisk config files
On 12-10-02 06:39 PM, Mitch Claborn wrote: Asterisk 1.8 on Ubuntu We store the configuration files in CVS. We have a development, QA and production environments. 90% of the config files are the same across all 3 environments, but there are some differences in sip.conf and extensions.conf (environment specific voip providers and/or analog/digital lines). I'd like to be able to use the same config files in CVS and have the differences resolved at run time, based on host name of the asterisk server. Any ideas how to do this? I looked at STS, but it appears to be Mac only. One idea would be to use something like #include sip-$$$hostname$$$.conf and then use sed or similar in the startup script to replace $$$hostname$$$ with the actual host name. Then each host/environment would have it's own include file as needed. Another idea would be to write a simple perl or other program to pre-process the files and put some markers in the files themselves. ; onlyif host=abc ; /onlyif The pre-processor would delete lines between the tags that didn't match the currently running host. If you are going to astricon you'll want to show up for my talk. This is basically what Leif and I will be talking about. I use puppet to help manage our 3 environments (test, stage and production). Along side it I use a the following configuration setup[1] plus some Debian packaging scripts[2]. With this, I can quickly spin up instances which are provisioned to a base. Then, depending on puppet manifests[3] for each node, it defines how the system is then provisioned. If more per-site settings are required, I'll roll them into Debian packages (we use Ubuntu 12.04) and have each site subscribe to a customer repo. [1] https://github.com/kickstandproject/asterisk/tree/master/debian/ast_config [2] https://github.com/kickstandproject/astricon-2012-presentation/tree/master/debian [3] https://github.com/kickstandproject/puppet-modules/tree/master/modules/asterisk/manifests -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
On 12-09-26 10:35 AM, motty.cruz wrote: Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! You are dialing a 8 digit number. Why? Also: Cause No. 27 - destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 1.8.15.0, Requested transfer capability: 0x00 - SPEECH
On 12-09-26 11:12 AM, motty.cruz wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Wednesday, September 26, 2012 7:52 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested transfer capability: 0x00 - SPEECH On 12-09-26 10:35 AM, motty.cruz wrote: Hello, I'm having issues connecting throu PRI with the following error Requested transfer capability: 0x00 - SPEECH Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660@voipphones:1] Set(SIP/4856-0003, CALLERID(num)=x) in new stack -- Executing [97052660@voipphones:2] Dial(SIP/4856-0003, dahdi/g1/97052660) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/97052660 -- Span 1: Channel 0/1 got hangup, cause 27 -- DAHDI/i1/97052660-4 is circuit-busy -- Hungup 'DAHDI/i1/97052660-4' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/4856-0003' status is 'CONGESTION' /etc/asterisk Chan_dahdi.conf [trunkgroups] [channels] ; PRI to Telco callerid=asreceived context=fromtelco switchtype=national signalling=pri_cpe group=1 channel = 1-23 ; pri to PBX context=frompbx switchtype=national signalling=pri_net group=2 channel = 25-47 In /etc/dahdi Modules Wct4xxp /etc/dahdi System.conf # PRI to Telco span=1,1,0,esf,b8zs bchan=1-23 dchan=24 # PRI to PBX span=2,0,0,esf,b8zs bchan=25-47 dchan=48 Any suggestoins are welcome! Thanks in advance! You are dialing a 8 digit number. Why? /* I'm dialing 8 digits because in my extensions.conf required user to dial 9 for outgoing calls. */ Right, but does your CO require you to pass the '9' to them or are you to strip it? Also: Cause No. 27 - destination out of order. This cause indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term not functioning correctly indicates that a signal message was unable to be delivered to the remote party; e.g., a physical layer or data link layer failure at the remote party or user equipment off-line. /* thanks for pointing that out, I overlook Cause No. 27. I will check aging my Dahdi configuration */ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Phone Configuration Overrides Not Saved
On 12-09-06 10:46 AM, Chris Nighswonger wrote: I have some Polycom 351 on Asterisk 10. On the same box as * I have a tftp server running to handle configs, etc. The Polycom phones have no problem grabbing config foo from the tftp server as well as writing log files back to the server. However, when I use the web-if on a phone to set a custom ring-tone, the web interface saves the change locally, but throws an error stating that it cannot write the override config to the tftp server. A look at a tcpdump shows that the phone indeed attempts to push the file to the server: 10:33:10.197294 IP 192.168.0.23.50672 pbx1.campus.foundations.edu.tftp: 38 RRQ 0004f2a5b892.cfg octet blksize 4096 10:33:10.473171 IP 192.168.0.23.29766 pbx1.campus.foundations.edu.tftp: 43 RRQ 2345-12365-001.sip.ld octet blksize 4096 10:33:10.711510 IP 192.168.0.23.33183 pbx1.campus.foundations.edu.tftp: 42 RRQ 0004f2a5b892_reg.cfg octet blksize 4096 10:33:10.758896 IP 192.168.0.23.28895 pbx1.campus.foundations.edu.tftp: 29 RRQ sip.cfg octet blksize 4096 10:33:11.620103 IP 192.168.0.23.21917 pbx1.campus.foundations.edu.tftp: 54 RRQ overrides/0004f2a5b892-phone.cfg octet blksize 4096 10:33:11.825803 IP 192.168.0.23.27460 pbx1.campus.foundations.edu.tftp: 52 RRQ overrides/0004f2a5b892-web.cfg octet blksize 4096 10:33:11.850646 IP 192.168.0.23.18554 pbx1.campus.foundations.edu.tftp: 55 RRQ licenses/-license.cfg octet blksize 4096 10:33:11.873077 IP 192.168.0.23.34766 pbx1.campus.foundations.edu.tftp: 55 RRQ licenses/0004f2a5b892-license.cfg octet blksize 4096 10:33:26.294928 IP 192.168.0.23.25504 pbx1.campus.foundations.edu.tftp: 37 WRQ cq_de_ku4dd.wav octet blksize 4096 10:33:36.238357 IP 192.168.0.23.63322 pbx1.campus.foundations.edu.tftp: 52 WRQ overrides/0004f2a5b892-web.cfg octet blksize 4096 10:33:36.539747 IP 192.168.0.23.37433 pbx1.campus.foundations.edu.tftp: 69 RRQ languages/Website_dictionary_language_en-us.xml octe A look at /var/log/messages shows: Sep 6 10:33:11 pbx1 in.tftpd[18368]: tftpd: read(ack): Connection refused Now why is it that the phone is refused only when writing the override file? Note that the only logging difference between a successful and unsuccessful write is the above line from the message log. The tcpdump looks the same. Permissions issues? If you switched to FTP or HTTP does it work? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 10 deb packages for Ubuntu 12.04?
On 12-09-03 07:46 AM, Stefan at WPF wrote: Hello, are there any deb packages for Ubuntu 12.04? The repos at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packagesare for older Ubuntu versions, also Asterisk 10 is only mentioned for YUM / CentOS? I'm working on a set for my personal repo, I'll share the link when finished -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 Package Question
On 12-08-28 10:14 PM, Chris Nighswonger wrote: Are there deb packages available for Asterisk 10 or for 11 beta? None. Well, the Debian VoIP team has an experimental[1] package for Asterisk 10. [1] http://anonscm.debian.org/viewvc/pkg-voip/asterisk/branches/experimental/ -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 community services - Old Mantis instance to be shutdown on Aug 28th, 2012
On 12-08-28 10:25 AM, Matthew Jordan wrote: - Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: asteriskt...@digium.com Sent: Monday, August 27, 2012 7:33:27 PM Subject: Re: [asterisk-users] Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012 Guys, Is it possible to leave the Mantis on permanently? It allows to productively search and work with issues recorded in it. Search, convenient straight forward layout, patch download URLs, everything just works there. JIRA maybe is convenient for the management and developers. I just guess, as somebody must have loved it so it was chosen as a Mantis replacement. But for an ordinary user (in my opinion) it is cumbersome and unfriendly. Sorry, but no. Infrastructure costs time, money, and effort by folks who have to keep the system running. You may not always see the problems - but they do happen, and Mantis certainly complicates all sorts of other pieces of the infrastructure puzzle. URLs that previously resolved to Mantis issues should automatically resolve to the JIRA issue that replaced them. I'm sorry you don't care for JIRA, but its the system we use now and its highly unlikely that we will migrate away from it anytime soon. I can understand Digium not wanting to spend time, money and effort, which is understandable. Thinking out loud, if the open source community still see value in the old mantis data, could we not hand it over for somebody else to manage? I'm sure we could find somebody to donate the bandwidth. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 1.8 and 11
On 12-08-22 02:04 PM, Giuseppe Longo wrote: Just a little questions, what's the difference between asterisk 1.8 and asterisk 11? Not a little answer[1]. [1] http://svnview.digium.com/svn/asterisk/branches/11/CHANGES?view=markup -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 1.8 and 11
On 12-08-22 03:47 PM, Danny Nicholas wrote: That's the theory. 11 is supposed to be 1.8 EOL version with some new tweaks. Keep in mind that 11 is officially a beta product, so if you're going to eat your own dog food 1.8 is probably the best option for now. Are you saying 1.8 is EOL? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] graceful restart
On 12-08-18 06:39 PM, Jan Blom wrote: Hello, Is there a way to detect, via cli or any other way, that Asterisk is in graceful shutdown mode, not accepting any new calls? Or to put the question a different way, how can I know that Asterisk has restarted again after the command core restart graceful in an automated way? Monitor the events on the AMI, you should see the following: Event: Shutdown Privilege: system,all SequenceNumber: 0 File: asterisk.c Line: 1773 Func: really_quit Shutdown: Cleanly Restart: True Then you can build out your monitoring tools from it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] RFC List
On 12-08-14 09:58 AM, Olle E. Johansson wrote: 8 aug 2012 kl. 14:07 skrev Kevin P. Fleming: On 08/08/2012 06:30 AM, Kannan wrote: Where can I get a complete set of RFCs and other specifications supported by Asterisk? To my knowledge there is no such list. In addition, Asterisk (like many other pieces of software) does not claim 100% compliance with every RFC that is relevant, so usually it's better to ask about the specific features you are interested in. THis is a document I haven't updated since 1.6.x but still covers a large part of the SIP implementation: http://svnview.digium.com/svn/asterisk/team/oej/sip-compliance/asterisk-sip.txt?view=markup Interesting, never knew this existed. I think it would be worth the time and effort to get this merged into trunk or into the wiki. A great piece of documentation. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Segmenting A Configration File
On 12-08-11 06:16 AM, Kannan wrote: Hi List, I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. I've been playing a lot with the following configuration layout[1], so far I am pretty happy how well it is working. We define the default values we want for the default configuration files, then add the site specific settings under the specific directory. Like I said, works very well for us. [1] https://github.com/kickstandproject/asterisk/tree/master/debian/ast_config -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Segmenting A Configration File
On 12-08-11 11:10 AM, Carlos Alvarez wrote: On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote: I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. We put each tenant's sip and extensions config files in /etc/asterisk/accounts and then do an include for that directory in the main files. We keep all the voicemail.conf in one because changes to passwords will NOT be saved to included files. We used to use includes for voicemail but that meant no password changes. This is no longer the case. Starting with 1.8 a new voicemail.conf setting (passwordlocation) has been added[1] to allow you to store the passwords outside the voicemail.conf file. With this setting the password gets written to secret.conf file within spooldir for each mailbox. That way, you can then breakout each mailbox into separate config files with include statements. [1] http://svnview.digium.com/svn/asterisk?revision=225406view=revision -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Debian 7/Asterisk TLS bug and others
On 12-08-10 04:47 PM, Daniel Pocock wrote: Debian 7 is currently in the `freeze' status with 1.8.13 - that means Debian 7 is very likely to release 1.8.13 and be carrying it for the next 2-3 years (typical lifetime of a Debian release) I run 1.8.8. TLS has a bug: it fails to receive BYE over the TLS connection from my Polycom phone. I tried 1.8.13, the version in Debian 7, and found a more severe bug: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=683956 The TLS clients can't connect at all, this looks like a really bad regression from 1.8.8 I've looked at 1.8.(14, 15, 16-rc1) and their changelogs don't mention any fix. Debian is very conservative about accepting updates during the `freeze' process - they will most likely want to see a 1.8.13.2 release with ONLY the most essential fixes a) is anyone else aware of these bugs? b) what essential changes should go into 1.8.13.2 for Debian? We don't need to release a 1.8.13.2 release of Asterisk. Once the issue has been fixed in the 1.8 release branch, it would just be back-ported into a Debian patch for the package. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Debian 7/Asterisk TLS bug and others
On 12-08-10 06:20 PM, Daniel Pocock wrote: Debian is very conservative about accepting updates during the `freeze' process - they will most likely want to see a 1.8.13.2 release with ONLY the most essential fixes a) is anyone else aware of these bugs? b) what essential changes should go into 1.8.13.2 for Debian? We don't need to release a 1.8.13.2 release of Asterisk. Once the issue has been fixed in the 1.8 release branch, it would just be back-ported into a Debian patch for the package. My impression was that a 1.8.13.2 release would be as conservative as any patches back-ported for the Debian package. It's not necessary, but it might be a convenient way to achieve the same goal. Is Digium officially endorsing 1.8.13 for wheezy in any way? No. Digium nor the Asterisk Project has anything to do with the package within Debian. In fact, most of the work is done by Tzafrir. Is anyone officially working on this particular problem already? I was tempted to have a closer look at it, but don't want to duplicate an effort that is already underway elsewhere. Best to check JIRA and see. Actually, does the issue even exist in JIRA? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 debian package and digium repository
On 12-08-07 03:31 AM, ml asterisk wrote: Hi, I used to install asterisk on debian squeeze with digium repository. The last build of asterisk available is 1.8.11.1. Is this repository discontinued ? Since leaving Digium they have become unmaintained. If you are interested in helping out, you might want to reach out to #asterisk-dev or asterisk-dev mailing list. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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
