[Asterisk-Users] Asterisk + G.729 on Sun T1000/T2000
I am curious if anyone has tested Asterisk running transcoding G.729 -> G.711 (ULAW) on Sun T1000 or T2000. I'd like to hear about your experience.ScriptHead ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk eating CPU
I have asterisk running user a user/group asterisk/asterisk like so su - asterisk safe_asterisk and one the processes utilizes way more CPU than any other. According to htop, it used 7:59:XX of CPU time. Once I kill asterisk and restart, another process does the same thing while others are running smoothly. This doesn't look like standard behavior to me. Is this some sort of a master process? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inform the agent about the queue he is answering
Yes, it is possible. You need to track the queue log and channels via manager console or by tailing logs in real time and then match the destination of the caller by the callerid. Then make the decision which URL to redirect the caller too. None of this comes with Asterisk but it is possible to build. On 2/3/06, nik600 <[EMAIL PROTECTED]> wrote: i'm planning to migrate a call center to asterisk, i don't understandif i can launch a resident application on the agent's client inrelation with the queue the agent's is answering.For example:I have - queue A- queue B- queue CAgent 100 (logged in A.B,C)Agent 101 (logged in C)When Agent 100 receives a call from the queue A i'd like to launch hisbrowser and point it to http://myserver/clientA, when the agentreceives a call from the queue B i'd link to launch his browser andpoint it to http://myserver/clientBIs it possible?With what soft-phone? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit sip sessions
You should create a secret dialing prefix like if you wanted to dial 1555333222 the user would actually have to dial 548261555333222. This way, even if they snatch the username/password but do not know the prefix, they won't be able to dial. On 2/2/06, Miguel <[EMAIL PROTECTED]> wrote: [EMAIL PROTECTED] wrote:>Shouldn't all sip users have different usernames?>(or am I missing some vital detail here?)>>PaulH>Yes Paul, Im in El Salvador and my users like to "share" their usernames/passwords and the original owner doesnt like to pay for callshe hasnt made.---Miguel___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] routing question: multipath routing for SIP
I have two T1s and I'd like to split my SIP traffic over the two. I am looking at this:http://lartc.org/howto/lartc.rpdb.multiple-links.html what bothers me about it is the note "Note that balancing will not be perfect, as it is route based, and routes are cached. This means that routes to often-used sites will always be over the same provider.". If all my traffic goes to the same IP, which is a remote SER proxy, will my second T1 be utilized at all? Does anyone have any experiece with this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT?: International number parsing
What you're trying to accomplish can be easily done with an SQL query. You need to create a table of all the prefixes (international dial+country code+city/carrier) and join by that prefix. On 1/27/06, Damon Estep <[EMAIL PROTECTED]> wrote: Can anyone shed some light on "rules" that might make the task ofparsing the country code and city codes from a dialed number in theCDRs?I know that there is almost never a case where a concatenated country and city code could overlap with another country code, but what aboutcity codes and local numbers? Is it possible for a concatenated citycode and local number to match another city code in the same country? I already have the table of country and city codes built.Are there holes in this theory;1. Starting after the international dialing code, find the longest matchfor country code.2. Starting after the country code from step 1, find the longest match for city code within that countries table of city codes.3. The rest is the local number.Are there known exceptions?Am I reinventing the wheel rather than finding the right alreadyexisting resource? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR reporting between two Asterisk servers
You have to write a CDR normalization script that would sift thru the calls and remove duplicate entries. It's also quite easy to do with a stored procedure, this way every time CDR gets written, it eliminates a duplicate. On 1/27/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Damon Estep wrote:> Use cdr_mysql>> Log your CDRs to a common database > Query as needed from either server using realtime() or from an external> app>Yeah, I thought about that. If it works how I think it wouldwork though I would have two CDR records for one call though. I would have one record from the remote server and one fromthe local. Correlating one record with another could be a pain.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fast AGI Options. Eeeek!
On 1/25/06, Sig Lange <[EMAIL PROTECTED]> wrote: On 1/25/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: Some questions regarding calling Fast AGI from the dial plan.Considering that the server side of the Fast AGI has to be able to a) use threading and b) connect to MySQL, this causes some serious limitations. I'm not a C programmer, so development options are either perl or python. In ref to using perl, You can't connect to the database for each thread? you would have to connect to the database from each thread, i strongly suggest you use something like sqlproxy (for mysql) or pgpool for postgres and actually have a thread close the connection once it's done, otherwise you're going to have a very large number of idle connections to the database. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to have a phone ring another extension as soon as off-hook?
I am seeking to implement the following behavor:When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement?ScriptHead ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] channel monitoring whisper mode?
As this isn't a part of *, has anyone accompilished a whisper mode in yet? What I am looking for is an ability for to say something while monitoring a channel and the agent being able to hear what I say while the called party is not. ScriptHead ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failover Registration
You can simply put then in order in your dial plan: exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED] exten => _1NXXNXX,2,Dial(SIP/[EMAIL PROTECTED] and so on. if carrier1 returns an error, * will dial out using carrier2. SciptHead On 12/2/05, Max Clark <[EMAIL PROTECTED]> wrote: Hi all,I would like to have two asterisk servers in a "cluster". From what Iunderstand using a mysql database I can store all of my peer/userinformation in the db and share this between servers. I can then take my polycom phone and register it to both of the asterisk servers at thesame time - so if one were to go offline traffic would be redirected tothe second.This works in theory for the end user - but how do I provide redundancy with my upstream providers? I.e. how do I fail over my registration toan upstream sip provider?Thanks in advance,Max-- Max Clark max [at] clarksys.com http://www.clarksys.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk cluster and astdb
This is correct, astdb isn't stored in a database and essentially your scenario with astdb file being shared won't work or will be highly unreliable. I am almost sure there was a patch in circulation, alas I didn't find it in the bug tracker. On 12/1/05, Bruce Ferrell <[EMAIL PROTECTED]> wrote: Matt Riddell wrote:> dashy dude wrote:>>>Dear All>>I am trying to build a high availability cluster of>>asterisk.>>I am using RedHat cluster management suit on>>Enterprise edition AS3 Origianally, astdb was located on native hard disk of>>each server.>>All my end points are configured for Reinvite=YesEverrything was working fine and if active server is >>rebooted, the standby would take over and the ongoing>>calls will continue without any problem.But this had a problem that the astdb file is not>>updated with latest end-point information and phones >>dont get a call untill they re register.To avoid this, I moved the astdb file on the shared>>storage and created sym links from individual servers.>>Now, when the active server is rebooted, all the >>active calls are dropped.Please help me in resolving this.>>> Why don't you use Asterisk RealTime?>correct if I'm wrong (frequently) but the call state isn't stored in the realtime db is it?linux-ha uses a form of shared disk called DRBD that might solve this ifyou forced the astdb onto that. Only one node of the cluster iscurrently allowed to write to astdb on that though. Just a thought___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom backlight?
I have actually spoken to a Polycom product engineer at VON and brought this up. It seems like it's a frequent request, hopefully they will address it soon. On 12/2/05, Wilson Pickett <[EMAIL PROTECTED]> wrote: Official Polycom view seems to be that you shouldn't work at night :)The phones are crying out for a backlit LCD that only lights whenambient light is low. I have a cheap radio/weather station with alarge LCD that does that. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A rather big setup.
Spreading * servers across multiple floors sounds like a bad idea since it'd increase maintenance time. With your projected call volume there's no way you can reliably run g729 or any other CPU hog of a codec on a single box. For this kind of a setup you'd need 2-3 boxes and a SER/heartbeat box to handle registration and call distribution. I would also isolate CDR recording to a separate box running a database like Postgres (IMHO better choice due to WAL) or MySQL. ScriptHead On 11/27/05, Simone Cittadini <[EMAIL PROTECTED]> wrote: Vedran Dakic ha scritto:>>I can only guess that I should have the ability to deliver a solution that>can do some 100/500 simultaneously. The only question is how powerful should>be a machine (or machines) that could do around 100/500 simultaneously. And, >just for the sake of knowing, what should the setup be alike if it was>240/1000 simultaneously?>>>My suggestion is to buy the E1 cards first of all and put them in a testserver, equipped with asterisk and all the relevant agi / db connections / moh etc..Then loop the card with a crossover cable and run some test script togenerate the medium and upper bound call flows.That should give you an idea of your cpu/ram requirements. >>In the second case there's no need for a cluster, a good server will do,>>(obviously a second server for backup is a good idea ). I'm assuming you>>can use a/ulaw to transmit the data, if bandwidth is a problem and you >>must compress cpu usage becomes a boottleneck to keep in mind.>>A/ulaw? I saw some reports that G.729 uses very little bandwidth and has>a quality part granted (audio quality). It's not a question of hardware >and/or CPU power, I have two dual Opteron configurations and could install>some more, it's just the question of that setup running with quality audio>and no unwanted events.>>G729 has a very good quality -considered the bandwidth used-, but if your customers are used to conventional telephony they will no doubtnotice the difference, so go with G711 (probably alaw, since you use E1I suppose you are in europe)Anyway if bandwidth is a problem consider ilbc / speex which are free and have good audio qualities also.Lastly a lot of the quality comes from a well configured phone, tweakwith volumes and timeouts.>I presume that I should have all of the phones using the same codec (so, >no transcoding), and preferrably the same VoIP protocol. I have a choice>there so everything's possible. Let's say - IP10s has H.323, SIP and MGCP>firmwares, although I'd like to leave H.323 out of the story. >>>Yes, leaving H323 out of the story is a good way to start the project :)___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to originate a call and capture it's DIALSTATUS
Hello, I've been trying to originate calls and capture the DIALSTAUS via the manager API. The problem seems that the API doesn't expose enough data to make a decision of what exactly happened to the call. It results in something like this: Action: Originate Channel: IAX2/switch/1 MaxRetries: 0 WaitTime: 2 Context: reminder Extension: s Priority: 1 Callerid: "Reminder" <555-555-> Event: Hangup Privilege: call,all Channel: IAX2/switch-3 Uniqueid: 1132271784.42 Cause: 0 Cause-txt: Unknown this is far from detailed. Is there a way to extract the actual DIALSTATUS such as ANSWER,BUSY,CONGESION, etc? The Cause doesn't seem to return 0 when the call is terminted thru IAX2 or SIP. It seems that it works on ZAP only. ScriptHead ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk-RealTime: sip_friends and register => user:[EMAIL PROTECTED]
I am upgrading to Asterisk-Realtime and stumbled upon a problem converting my existing sip.conf register command to the RealTime format. It seems that sip_friends table setup doesn't allow for such thing to happen. So far the only way I see to do this is dumping the sip_friends table setup in favor of Asterisk RealTime Static (http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static) which seems to be quite an ugly solution. Am I missing anything? ScriptHead ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanSpy and Sipura 2100 jitter.
I have an analog phone connected to a Sipura 2100 which in turn connecteds to * over a 100mbps LAN. When I do ChanSpy on a bridged call, it causes massive jitter. When I attempt ChanSpy with a Grandstream GXP-2000 the monitored call is clear. Has anyone had this happen? Any suggestions? ScriptHead ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo cancellation on an iax2 channel
I have minor echo on an IAX2 channel when using Firefly and a head set. I have tried various headsets and settings but still a little bit of the echo remains and I'd love to get rid of it. After some research I stubled on zaptel/mec2.h but it seem that it works only on the ZAP channel. Is there something I can do on the IAX2 channel? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice-coloring with asterisk
I was pondering of the best way to implement voice-coloring within Asterisk, e.g. pass a channel thru a multiband equalizer and modify it enough where it could be distinguished from other voices in a conference call. This could make conference calls much less confusing. Perhaps the easiest way would be to use sox as the equalizer but I am not familiar enough with * to know how to put a channel thru sox. Anyone? Scripthead ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_pgsql amaflags are always 3
I use cdr_pgsql on asterisk-head to store my CDR records. The issue is that regardless to what I set amaflags to in iax.conf, the data that gets written to the cdr.amaflags is always 3. What am I missing? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
