Re: [asterisk-users] GUI for Asterisk
If you plan it right from the start, FreePBX can save hell lot of time. Instead of fixing in include files, you can also create custom contexts from within the GUI now, i am sure there is a module for that as well. As said above, either stick fully to GUI or fully to manual configurations. Ugly mixing of the two will definitely bite you later on if you don't know what you are doing . On Thu, Jun 25, 2009 at 2:41 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Jun 25, 2009 at 11:02:44AM +0200, jonas kellens wrote: Tzafrir Cohen, if mixing hand-written configs with GUI-configs is not 'good practise', then how to build a scalable Asterisk IP-PBX where the customer is not 100% dependent of the implementer ? Like I already said, I got the remark To add a new phone, I do not want to be forced to call you. And I don't see a CEO of a meat-company learning some vim-skills... I don't know how to put the simpler administration into the hands of a noob, without me having to put a 100% support into the contract (which is overkill). It means you should adapt the said GUI to generate the right configuration. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialer program
There is also GNUdial but i would prefer VICIdial anyday over it ( personal opinion :) ) . On Wed, Jun 10, 2009 at 9:43 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank you Jose. Interesting suggestion! Is there any other? On Wed, Jun 10, 2009 at 11:21 AM, Jose P. Espinal j...@slackware-es.comwrote: Hola Carlos, Have you searched for ViciDialer? It's a good one. Give it a shot, it might be what you are looking for. Carlos Ruiz Diaz wrote: Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my time in something that nobody is going to use. Do you people think that a dialer could be considered a successful project? Thanks in advance. Carlos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hacked
Here's what fail2ban service caught The IP 89.111.184.221 has just been banned by Fail2Ban after 80 attempts against ASTERISK. On Wed, Apr 8, 2009 at 7:01 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 07 April 2009 11:28:52 Tilghman Lesher wrote: The recent vulnerability had nothing to do with this, but with the ability of an attacker to scan a SIP server for legitimate usernames and passwords. This, by the way, merely took advantage of the SIP protocol, as written. Normally, SIP allows you to differentiate between invalid usernames (404) and invalid passwords (403). What we closed in the recent vulnerability patch was to allow administrators to send back 403, regardless of whether the username existed or not. By the way, I am VASTLY oversimplifying the return codes here for the sake of clarity. The actual return code is based upon a number of factors, but it is modeled to return the same responses as would a bad password with a legitimate user account (thus making it impossible, externally, to tell the difference between a legitimate user account and a non-existent user account). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPkall
I registered few days back and got a DID. Maybe this is temporary ? On Mon, Apr 6, 2009 at 7:05 PM, Steve Howes st...@geekinter.net wrote: On 6 Apr 2009, at 14:32, Dean Collins wrote: None of their pages apart from the front page seem to work though http://phone.ipkall.com/ipphone/login.asp http://phone.ipkall.com/login.asp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
[442033553] user=442033553 type=pusers secret=1234 host=dynamic context=users nat=yes make it context=stations , i am assuming this is how your DID provider is sending u calls ? Let us know if your DID provider is just sending calls to your ip address or you are registering asterisk server with the, . Keep context=stations in extensions.conf global section . On Thu, Sep 4, 2008 at 2:41 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Hey, Did you reload asterisk after changing the extensions.conf? Also, if you try it with sip set debug on the console what do you see? michel freiha wrote: Hello Air, I did what you asked for but I got the following error: extensions.conf: [stations] exten = 442033553,1,Answer exten = 442033553,n,Playback(demo-nogo) Error message: [Sep 3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found. Regards On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: michel freiha wrote: Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me some tips about how to accomplish this task? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, I have never used that provider but usually either the provider knows your switch's ip and routes the did traffic to it or you have asterisk register with the provider so that it knows where to route the calls. Once thats done you can do something like exten = XX,1,Answer exten = XX,n,Playback(file) Where the x's are the number that you see coming in from your provider. If you're routed all your dids from what looks like one number(callcentric does this) then you might need to use the sip header to route your did to the particular extension you want. You shouldn't have to bother with this if you only have one did. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com http://www.escapetel.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unable to register to tnet.it
Check dns server entries in asterisk box . /etc/resolv.conf . Put opendns servers ip there just to test . opendns ip's are 208.67.220.220 and 208.67.222.222 On Tue, Jul 15, 2008 at 2:19 PM, map [EMAIL PROTECTED] wrote: Hi Giorgio, RE my point 2: You should test a sip client, whatever you want, on your linux/asterisk box just to double check that this box works fine. If you are abel to connect with a sip client from tour asterisk box we will be sure that the network configuration is ok. You have no natt but maybe your routing table is not correct :-) Do you already test to just ping to tnet.it port 5060 ? Marino On Tue, Jul 15, 2008 at 10:27 AM, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Marino, 1) yes I can connect using the account 2) no, I'm running zoiper on a different machine. I'm using an Asterisk server which is not behind nat as for the machine zoiper is runnin' on. The Asterisk server is directly connected to internet, I wanted to avoid nat problems, that's why. Moreover I tried to create a simpler account on my zoiper using username, password and domain name only and it works even without setting the sip proxy. I changed the Asterisk server too: now I'm using a test one where I can ping tnet.it from... but nothing changes. I'm using this string: register = 0442410280:provapolika:[EMAIL PROTECTED]/0442410280 I changed it in many other forms following the wiki pages but nothing. I see sip packets are sent to tnet.it (I set up sip debug) but I always get this message: Jul 15 10:06:39 NOTICE[3281]: chan_sip.c:5495 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1) I wonder why I had no problems with the other provider we are using while tnet.it is making me get crazy Thank you. Giorgio map wrote: Hi Giorgio, Just to recap: 1) you are able to connect to tnet.it http://tnet.it by using the same account of your asterisk box. There is no issue related to your account. 2) Could you please confirm that you are running zoiper from the same box used by asterisk? If yes we can exclude some generic network issues. From your previous email : ... Activating sip debug shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named Ethereal in order to check sip messages. I have to sniff both asterisk and zoiper sip messages. I know that this can be tricky but this can help you to understand what is wrong in sip messages. Please let me know if you need more detail. Marino On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, I tried to connect zoiper directly to the provider with the same account parameters I'm using with Asterisk. Zoiper connects without problems. It is true tnet.it http://tnet.it is not resolvable but I can use the proxy URL sip.tnet.it http://sip.tnet.it which seems to work with Zoiper but not with Asterisk. I'm trying to understand where is the problem. I thought I had to specify the outboundproxy parameter in the general section of sip.conf to make Asterisk correctly work but it seems that's not enough. Thank you. Giorgio map wrote: Hi Giorgio, From your email seems clear that your Asterisk box can not resolve tnet.it http://tnet.it http://tnet.it and SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Marino, Asterisk gives a timeout on registration and a no such host because cannot resolve tnet.it http://tnet.it http://tnet.it but that server address is not resolvable so I think that is not a problem (my zoiper connects to the provider without problems, so why shouldn't Asterisk??) Activating sip debug shows the register packets but nothing in return. I used the proxy tnet gave me but nothing changes. Searched on their site for some help about Asterisk
Re: [asterisk-users] g729 encoder/decoder
When g729 phone calls another g729 phone and you are not recording calls or doing meetme with them then license is not required ... g729 phone calling g711 will require a license to transcode the g729 side ( no license for g711 side of call ) . In short anytime u need to convert g729 into some other codec ( transcoding ) you need 1 license . On Wed, Apr 2, 2008 at 1:59 AM, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: How does the g729 encoder/decoder count in regards to the total number of licenses and how does it count an encoder/decoder? I looked on the wiki and don't really see anything that explains it. In other words, how do the calls below count (assume no reinvite)? g729 phone calls into voicemail g729 phone calls g711 phone g729 phone calls other g729 phone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G723 on asterisk 1.4.1
That's strange , i am able to see the *url* in Martin's reply . On Sun, Mar 23, 2008 at 6:14 PM, Eric Wieling [EMAIL PROTECTED] wrote: The only messages I have EVER seen Digium remove from the mailing list archives are discussions about this unlicensed codec. Martin wrote: Download an appropriate binary from [url removed] and just drop into /usr/lib/asterisk/modules/ add allow=g723 to your sip.conf as necessary and restart asterisk... Im only not sure how legal is this, you will probably need to obtain licenses for all concurent channels... Martin - Original Message - From: wassim darwish [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 22. brezna 2008 15:21 Subject: [asterisk-users] G723 on asterisk 1.4.1 Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk re-invites and billing
I am using asterisk 1.4.18 (server A ) and have it store records in mysql database . One of my client uses predictive dialer ( asterisk 1.2.26 based and server B ) which makes many calls . B registers with A over sip and there is no nat involved If i re-invite rtp from server B to my carrier ( server A in between ) I saw many calls having duration of 0,1 or 2 seconds on server A's cdr but surprisingly all these calls were marked at 15 minutes usage on my provider's records . My sip route provider himself is re-inviting traffic ahead to their media gateways . I have gone through asterisk sip.conf and i don't see any setting limiting anything to around 15 minutes , default rtp timeout settings are around 60 seconds in asterisk . My provider says that they don't have any 15 minute limit on their end . The records on server B also suggests that calls are indeed very small 1 to- 3 seconds . Server A and B both have static ip's and there is no bandwith problem on server A . If i disable re invites on server A then this problem isn't present . Did anybody else have this kind of problem ? Any suggestions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not transcoding between installed codecs
iax.conf doesn't take canreinvite=no . It's only for sip . For iax2 its transfer=yes/no (1.4 ) or notransfer=yes (1.2) , there's one more parameter in 1.4 with which u can transfer only audio stream . Check voip-info wiki for all options . On Thu, Mar 13, 2008 at 7:48 AM, Gonzalo Servat [EMAIL PROTECTED] wrote: On Wed, Mar 12, 2008 at 12:58 PM, Brent Davidson [EMAIL PROTECTED] wrote: Do you have canreinvite=no in the sip client configuration? If not then the two sip phones are probably issuing a reinvite command and taking asterisk out of the call path. If that happens and the phones can't reach consensus on a codec then you run into audio problems. If you're not a provider and just using asterisk as a PBX then it's probably better to set the phones up with a matching codec set and allow them to establish a direct connection between each other to keep load off the Asterisk server. Otherwise set canreinvite=no and Asterisk should transcode correctly. Brent, Thank you vry much for replying. I thought the message went unseen but found your reply when I went to look at the thread :) You're absolutely right. Looks like the SIP client was messing up (or something) when different codecs were used. I tried canreinvite=no and it worked perfectly, but as you said, it's best to bypass Asterisk when talking between clients on the same network. I tried a different IAX client and it had no problems using different codecs (with canreinvite=yes) so all is good. Thanks again! Gonzalo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme voice quality problems
Ubuntu has a real time kernel in repository apt-get install linux-rt . So you dont need to recompile . I think debian should also have one in repository , or u can manually compile a real time enabled kernel . Here's what is shows with real time patched kernel . dmesg|grep ztdummy [ 53.293071] ztdummy: Trying to load High Resolution Timer [ 53.293076] ztdummy: Initialized High Resolution Timer [ 53.293078] ztdummy: Starting High Resolution Timer [ 53.293080] ztdummy: High Resolution Timer started, good to go zttest Opened pseudo zap interface, measuring accuracy... 100.00% 99.987793% 99.792480% 99.780273% 99.987793% 99.975586% 99.987793% 99.987793% 100.00% 100.00% 100.00% 99.987793% 99.987793% 99.987793% 100.00% 100.00% 99.987793% 100.00% --- Results after 18 passes --- Best: 100.00 -- Worst: 99.780273 -- Average: 99.969482 On Feb 2, 2008 10:27 PM, Administrator TOOTAI [EMAIL PROTECTED] wrote: Matthew J. Roth a écrit : Administrator TOOTAI wrote: This is not true if you're using B410P cards. We always face timing problem as we can't -Asterisk stability issues- add X100P or TDM400P with those cards Daniel, I thought that using an empty TDM400P as a timing source may no longer be the best solution due to the emergence of new stable timing sources (such as HPET), but this is an interesting issue. Are you stating that you can't put an X100P or a TDM400P with no lines attached alongside a B410P because it impacts the stability of Asterisk? Yes Do you have any idea why? No Can't the B410P be used as a timing source? No What have you done to provide stable timing? ztdummy, not always stable :-( I know that's a lot of questions, but I'm genuinely curious. ;-) It seems very strange that a TDM400P in timingonly mode and no lines attached would have any impact on Asterisk's stability. I have to add that this is mainly true with 2 B410P in the server or with B410P in amd64 machines. Seems also that Debian Etch with a 2.6.18 kernel is not the best :-( -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Enterprise or Fedora?
I prefer CentOS barebone install and yumming the way up for dependencies but manually compile asterisk/zaptel . Ubuntu servers are pretty good too since its repositories are quite bigger compared to CentOS . On Feb 2, 2008 11:45 PM, shadowym [EMAIL PROTECTED] wrote: Actively maintained or actively being broken and fixed with constant updates? Not something suitable for Production IMHO. Makes more sense for development and experimentation IMHO. -Original Message- From: Benny Amorsen [mailto:[EMAIL PROTECTED] Sent: Friday, February 01, 2008 1:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Enterprise or Fedora? shadowym [EMAIL PROTECTED] writes: I cannot think of a single reason to use Fedora for a production anything when there are alternatives like CentOS. Fedora is bleeding edge stuff and constantly changing. The advantage of Fedora is that it is very actively maintained -- and asterisk is only a yum install asterisk away! /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disable IAX2 call path optimization
You are usinfg sip or iax ? Its possible to prevent in both cases for sip under peer definition you can put canreinvite=no and in iax2 you can put transfer=no for asterisk 1.4 or notranser=yes in asterisk 1.2 .. Search for this on voip-info.org wiki for more info . On Jan 25, 2008 7:03 PM, [EMAIL PROTECTED] wrote: I have a call coming in from Asterisk-A going to Asterisk-B where it's determined that the called party is in fact yet another number in Asterisk-A so a new call is created from B to A and the two calls bridged (by Asterisk) at Asterisk-B. Originating Caller == Asterisk-A == Asterisk-B == Asterisk-A Now, what happens is that in my case both A and B are on the same network and therefore Asterisk-A apparently optimizes the round-trip to Asterisk-B out and the original caller talks directly to the extension hosted in Asterisk-A without the call path going the round-trip to Asterisk-B. Is it possible to prevent this optimization from happening? Any way to control if it happens at all, or can it be selected on per-call basis somehow? Can I find anywhere more details of call path optimization and it's configuration, use, functionality and behaviour? tnx, Baldvin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Your favorite Asterisk application.
I like the Echo application in asterisk ;) . Weird :P On Jan 24, 2008 7:07 PM, Mark Johnson [EMAIL PROTECTED] wrote: Ken D'Ambrosio wrote: Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken We moved from a Cisco Call Manager about 2.5 years ago to Asterisk. One of the hurdles I had was that the Call Manager had a receptionist panel so they could see who was on the phone, transfer calls, etc... I set up a demo of of the Flash Operator Panel and it alleviated that sticking point. It's a little slower than an executable would be, but it's web based and flash so it's runs on just about every browser and OS. You can even do some slick things like pop up windows in the browser to provide information about who is calling. Works good for a CMS system where a customer service rep can automatically be shown information about the customer who is on the line. http://www.asternic.org/ -- Mark Johnson http://www.astroshapes.com/information-technology/blog/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime problem host='dynamic' in 1.2.26.1
Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Quite obvious .. doest sippeers have that row ? On Jan 24, 2008 6:04 PM, Torbjörn Abrahamsson [EMAIL PROTECTED] wrote: Developers and maintainers, any information? // T Torbjörn Abrahamsson wrote: Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial(SIP/dev02-08c36f28, SIP/[EMAIL PROTECTED]||W) in new stack Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Everything is fine. Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic' Jan 23 09:02:07 WARNING[2236] chan_sip.c: No such host: 989800-out Jan 23 09:02:07 NOTICE[2236] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Jan 23 09:02:07 VERBOSE[2236] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Jan 23 09:02:07 DEBUG[2236] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. I looked in the archives and found this thread: http://lists.digium.com/pipermail/asterisk-users/2007-December/202616.html Here the same problem is discussed for the 1.4 branch, and the result is that the problem should be fixed. But this is still a problem in 1.2branch. Will this be corrected in a new release, or is this not considered a security fix and hence ignored? Actually isn't this a fix for a security fix... BR, Torbjörn Abrahamsson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call-limit in database
call-limit is to set number of alternate calls . and L is to limit duration of each call . On Dec 22, 2007 2:54 PM, Pezhman Lali [EMAIL PROTECTED] wrote: Dear I am using this function with L for example in the dbase. app=Dial appdata=SIP/[EMAIL PROTECTED]|60|L(10) it means dial 1 thru 1.1.1.1, with limitation=10 mili-second, and time out=60 sec best Mani --- Bhrugu Mehta [EMAIL PROTECTED] wrote: hi, all proble: I have add CALL-LIMIT field in my sip table in mysql. but when i call using sip same error occurred when use simple sip.conf file. is this possible to add CALL-LIMIT field in sip realtime table in mysql. if yes than how Bhrugu Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP call interrupted after 64 seconds
Can you post the part of your dialplan which causes this behaviour ? On Dec 17, 2007 11:19 PM, Roger Schreiter [EMAIL PROTECTED] wrote: Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to asterisk-1.4.8, and do encounter the same problem. I have other asterisk machines running, using the same dialplan, without this problem. Did anyone else observe this strange behaviour of calls ending after 64 secondes of uptime? My os is Suse-Linux 10.2. Thanks for any hints! Roger. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP port 5060 closed - how do I open it?
Its pretty clear from netstat that asterisk is listening on udp 5060 . It might be firewall configuration in server thats blocking it . Also you might have scanned for TCP port 5060 from outside and hence u find it closed ? On Nov 28, 2007 5:57 AM, Nick Brown [EMAIL PROTECTED] wrote: Zaheer, On 28/11/07 9:28 AM, Zaheer K. Master [EMAIL PROTECTED] wrote: Yes I have a sip.conf, contents as follows: From the CLI can you confirm SIP is running by pasting the results of 'module show like sip' Cheers Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
asterisk -rx module load codec_g729.so or module load codec_g729 and shhow translation recalc On Nov 27, 2007 2:36 AM, Fernando Berretta [EMAIL PROTECTED] wrote: Dear Mindaugas, Thanks for your promt response I've already tried this but.. it's not working,, what file do you think I should use ? any other idea ? Best Regards, Fernando Mindaugas Kezys wrote: Rename to codec_g729.sohttp://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so Copy to /usr/lib/asterisk/modules chmod 777 codec_g729.so restart Asterisk show translations Mindaugas Kezys http://www.kolmisoft.com Advanced Billing for Asterisk PBX *From:* [EMAIL PROTECTED] [ mailto:[EMAIL PROTECTED][EMAIL PROTECTED]] *On Behalf Of *Fernando Berretta *Sent:* Monday, November 26, 2007 6:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Dear Mindaugas, I've already download the folowing files for testing codec_g729-ast14-gcc4-glibc-athlon-sse.sohttp://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so codec_g729-ast14-gcc4-glibc-core2.sohttp://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-core2.so codec_g729-ast14-icc-glibc-x86_64-core2.sohttp://asterisk.hosting.lv/bin/codec_g729-ast14-icc-glibc-x86_64-core2.so But... no one of them seems to be working -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
Well one of them should work fine ;) . I was not sure if it required .so extension ( i guess it doesnt ) anyway hitting tab can autocomplete or atleast give hints . Looks like he is loading wrong module bcoz asterisk autoloads this on restart if placed in proper directory with proper permissions :) . On Nov 27, 2007 3:33 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Nov 27, 2007 at 03:00:19AM +0530, Jaswinder Singh wrote: asterisk -rx module load codec_g729.so or module load codec_g729 and shhow translation recalc You ca't really fix a typo without intrducing a new one, eh? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite
Latest version of X-lite does not have GSM codec . Downgrde your versiona dn you will get gsm codec . I read on their forums that next version will again be including GSM codec . On 03/11/2007, Julio Tejera [EMAIL PROTECTED] wrote: Latest version of X-Lite does not support GSM codecs any more It could be a good idea that you post on the rigth place not here :o) jat - Original Message - From: Alejandro Cabrera Obed [EMAIL PROTECTED] To: asterisk Users Mailing List asterisk-users@lists.digium.com Sent: Friday, November 02, 2007 2:05 PM Subject: Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite SIP wrote: Alejandro Cabrera Obed wrote: Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in allow=gsm line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF manual of X-Lite 3.0 say it has GSM builtin support. Do you know what's the matter with X-Lite and GSM ??? Can I add it ??? Really thanks Alejandro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It lists GSM on my audio codec settings. Perhaps there's something wrong with your install? Try disabling the Zero Touch bandwidth detection. It has, in the past, interfered with my selection of codecs. N. Thanks for your support...I've uninstalled my X-Lite 3.0 softphone and after that I've downloaded the X-Lite 3.0 again from the official web site. But when I go to audio codecs settings, the GSM codec is not present. I disable the zero touch bandwith detection and restart the softphone, but the GSM codec is not present at all. Any idea ??? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Internal Virus Database is out-of-date. Checked by AVG. Version: 7.5.446 / Virus Database: 269.3.0/758 - Release Date: 4/12/2007 11:52 AM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk.conf and it's impact on CLI
astrundir = /var/run Change this to astrundir = /var/run/asterisk on 1.4 server and chmod /var/run/asterisk to 777 . make sure u create that directory as well . On 20/10/2007, Al lists [EMAIL PROTECTED] wrote: this message is basically tells you asterisk is not running. can you check and see if asterisk is running and present in memory? something like ps -ef | grep asterisk On 10/20/07, Dominic Son [EMAIL PROTECTED] wrote: I was previous using Asterisk 1.2.9.1 and decided to get some real servers outside of my house. It was time for Asterisk 1.4.4. I figured since all the conf files were in /etc/asterisk form the old box, i'd just copy tha directory over to the new server. My SIP DID AGI stuff worked, except running 'asterisk -r' doesn't. It tells me ' Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)' Basically, the difference between 'asterisk.conf' file is as follows: v 1.2.9 (installed through trixbox) astrundir = /var/run/asterisk v 1.4.4 astrundir = /var/run So in my new servers, if i keep it as '/var/run/asterisk, my DID phone will work with stanaphone (in which i'm crapping in my pants if they'll exist cause they never return emails). Though CLI won't work. if i do '/var/run', my DID won't work, but CLI will... I've tried just coping over the extensions_additional.conf and sip_additional.conf files from my old setup to my new one, and that didn't work. Maybe I should just install my previous version. Are there QoS differences though? I'd rather not regress if that were the case. -- Anything else, let me know. - Dominic It is not the force of a stroke that makes fine art ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Injecting a sound file into a bridged call
See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk to one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . But i dont know how to play a recorded file in it . On 08/10/2007, Girts Graudins [EMAIL PROTECTED] wrote: Hello everyone, I'm looking for a way to play a sound file to an already established bridged call. It is meant for one party, but it's ok if both parties would hear it. Ideally, I'd like to be able to trigger this from the Management Interface with something like: Action: Playback File: tt-weasels Channel: Zap/nn However, I haven't seen anything like that being available, so I'm looking for other suggestions. The critical pieces are as follows: 1) I need to be able to initiate this as an outside event/command; like I said, MI would be ideal; 2) I've seen whisper-type of functionality associated with meetme rooms, but I'd rather not set up a dynamic meetme room for each call I'm bridging; 3) Obviously there's Playback() and Background() available in the dialplan, but I need to be able to trigger the sound at will after the call's already been established. This sounds like a simple thing to wish for, yet I don't see a ready answer. Any tips would be appreciated. TIA, Girts ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo problems
Also many people using softphone turn's on mic boost in windows xp which also makes echo if it is set to very loud . On 30/09/2007, Philipp Kempgen [EMAIL PROTECTED] wrote: http://linux.sgms-centre.com/misc/netiquette.php#threading http://linux.sgms-centre.com/misc/netiquette.php#toppost SCNR Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Home system with SIP
since asterisk is only using operating system's routing ability , you can always set static routes using route command in linux . On 26/09/2007, Jeremy Mann [EMAIL PROTECTED] wrote: Why did you waste time with this reply? You do realize some users don't have control over their Exchange servers, and asinine footers are placed into an email without their intervention or control right? -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Benny Amorsen Sent: Tuesday, September 25, 2007 1:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Multiple Home system with SIP JM == Jeremy Mann [EMAIL PROTECTED] writes: I would have answered, but I was prohibited from quoting properly: JM If you are the intended recipient, further disclosures are JM prohibited without proper authorization. /Benny This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] errors messages in asterisk CLI
Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away This part is more like mistake in /etc/asterisk/cdr_mysql.conf . Check it once and relaod asterisk , then you can type cdr mysql status in cli to check if it connects to mysql properly . Sep 21 20:31:49 ERROR[3774]: cdr_csv.c:237 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied Permission error try chowning the directory to user which asterisk runs on OR chmod 777 /var/log/asterisk/* -R Sep 21 20:31:49 ERROR[3774]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied Same permission error . On 22/09/2007, Jody Gugelhupf [EMAIL PROTECTED] wrote: hi ppl, i have a problem, i get these messages in the asterisk CLI: Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away Sep 21 20:31:49 ERROR[3774]: cdr_csv.c:237 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied Sep 21 20:31:49 ERROR[3774]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied how can i fix these errors? here some info about my system: debian etch 4.0 kernel: 2.6.18-4-686 Asterisk 1.2.13 VoiceOne version is v. 0.5.0 using plugin subsystem v. 0.4pre3 mysql Ver 14.12 Distrib 5.0.32, for pc-linux-gnu (i486) using readline 5.2 PHP 4.4.4-8+etch4 (cli) (built: Jun 30 2007 21:02:54) using grandstream(handytone) 486 as sip device, no other devices or PSTN connected, only using sip/voip providers behind router/NAT thx in advance :) jody :) Yahoo! Canada Toolbar: Search from anywhere on the web, and bookmark your favourite sites. Download it now at http://ca.toolbar.yahoo.com. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] prepaid application recommendation
A2billing is very versatile and good solution for asterisk prepaid/postpaid billing . On 22/09/2007, Apa Minerala [EMAIL PROTECTED] wrote: You should make sure you know how to install it yourself. And you should also test it very very VERY carefully. I can't underline very enough. And if ever you ask for service, get a real company, with a real person behind the desk, who is doing only this. I have had my sad story with the A2Billing people. Tudor *Sarfaraz Chougule [EMAIL PROTECTED]* wrote: I would recomend using Areski's billing solution : http://www.areski.net/a2billing On 9/21/07, Rilawich Ango [EMAIL PROTECTED] wrote: Hi all, I am looking for a prepaid application. I found that there are many applications in the page http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications. Anyone recommendation among them? ango ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- With Best Regards, ** Sarfaraz Chougule ** ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Yahoo! oneSearch: Finally, mobile search that gives answershttp://us.rd.yahoo.com/evt=48252/*http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC, not web links. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP and Firewall
Here you go http://www.voip-info.org/wiki/view/Asterisk+firewall+rules . You can also set your rtp.conf properly and open very few rtp ports instead of all 1-2 udp ports . On 22/09/2007, Guenther Sohler [EMAIL PROTECTED] wrote: Hallo, I'd like to correctly set up my firewall in my system for udp and asterisk I have got a server, which has got one static ip adress to the internet. Asterisks is running on this server. It registers at sipgate.at and mujtelefon.com The Server also does nat to the my intranet, where my pc and my hardware sip phone sits. The Hardware sip phone registers to asterisk on my server from its intranet ip adress. Everything works fine. The question is just: How to code good stateful firewall rules with iptables and netfilter_sip ? What would be apropriate to my system ? rds -- GMX FreeMail: 1 GB Postfach, 5 E-Mail-Adressen, 10 Free SMS. Alle Infos und kostenlose Anmeldung: http://www.gmx.net/de/go/freemail ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?
I prefer centos , debian/ubuntu are also a good option . It just depends on which distribution you are comfortable with . We also have asterisk running very stable on slackware . On 12/09/2007, Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 12 Sep 2007, Euler Pereira wrote: Hey all! I'm newbie in the Asterisk World but old in other telephony systems like Lucent/Avaya, Sopho, Siemens and Linux/Unix system. I'm in doubt, as based system, should I install Fedora, Debian, Slackware, FreeBSD our Sun Solaris? Which is more robust for a small Asterisk system, about 8 extensions, 4 hardphone and 4 softphone? Which of Fedora, Debian or Slackware do you know best? I use Debian, but that's because it's the one I know best. Gordon ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference in show channels
'show channels' shows only running calls while 'sip show channels' shows all running sip sessions including phones trying to register . On 09/09/2007, ram [EMAIL PROTECTED] wrote: Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configure extension by software
You can use asterisk realtime which can read sip config from database ( mysql/pgsql) . Your application can just write info to database and asterisk will read it and make peers . You can also include a custom config file within sip.conf and make your application write peer settings to that file and reload asterisk by using asterisk management interface . On 08/09/2007, phananhvu [EMAIL PROTECTED] wrote: Before an IP Phones can be registered to an Asterisk server, the extension for it must be configured in Asterisk. Usually, Asterisk adminintor must add the extension by hand. Is there any library, API to do this by software??? For example, i want to develope a software that add new extensions to Asterisk system, sothat, any IP Phones can use that extensions to establish a call. I'm digging on Asterisk-Java but this library seems not support this. Anybody has dealed with this before ?? Phan Anh Vu DT12.K49.HUT RDLab ( C9.410 ) HUT -- Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge http://us.rd.yahoo.com/evt=47093/*http://tv.yahoo.com/collections/222to see what's on, when. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A102d sangoma's card and ztdummy
Sin you have sangoma card , it will act as timer . You need to install meetme ( app_conference is not very stable last time i read ) . On 01/09/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call service and I use A102d sangoma's card.Do I should install ztdummy or app-conference? Best regards. -- Yahoo! oneSearch: Finally, mobile search that gives answershttp://us.rd.yahoo.com/evt=48252/*http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC, not web links. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk multiport
What i actually do is make asterisk listen on some other port like 5097 and redirect port 5060 to it with iptables like this /sbin/iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5060 -j DNAT --to YOURIPHERE:5097 This works very well . If i make asterisk listen on 5060 and redirect say 5097 to 5060 i had lot of problems with firewalled systems ( blocked 5060 by isp ) . Also on blocked end its recommended to use some softphone like xlite which completely allows you to set custom ports on machine itself to listen, taking 5060 completely out of picture . On 17/08/07, Steve Totaro [EMAIL PROTECTED] wrote: Steven wrote: I am curious. Why would one need to do this? If a phone connect to 5060 from another port number, asterisk happily works, so why use multiple port on asterisk? I cannot see the thread history but from the context, I would say because many ISPs block 5060, 25, and others. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR billsec greater than duration
Yes it maybe a hung channel problem .. but question is no matter how much billsec ... should duration be more than that ? On 16/08/07, Anthony Francis [EMAIL PROTECTED] wrote: You are a victim of hung channels, just write a script that corrects this. Anthony Mail list wrote: The destination numbers are valid in almost all cases . But i do think it might be when someone is on call and on client side internet connection goes off .. I am really not sure about this one but i just saw that maximum such records are from one of my customer who has a very bad connection . On 16/08/07, *Edoardo Serra* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I noticed that fpbx calls ResetCDR on call hangup (don't know why this choice) Could it be related to that ?? Tnx E. Jaswinder Singh ha scritto: I made same thread few months ago and many people said that they dont have such records in plain asterisk install ( no freepbx ) . I was also using freepbx when i had this problem . Heres mine : mysql select count(*) from cdr where billsec duration; +--+ | count(*) | +--+ | 124 | +--+ this is out of 1749216 cdr records . I am also using freepbx btw . In all such cdr's duration is always 0 but billsec varies . On 15/08/07, *Edoardo Serra* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR billsec greater than duration
I made same thread few months ago and many people said that they dont have such records in plain asterisk install ( no freepbx ) . I was also using freepbx when i had this problem . Heres mine : mysql select count(*) from cdr where billsec duration; +--+ | count(*) | +--+ | 124 | +--+ this is out of 1749216 cdr records . I am also using freepbx btw . In all such cdr's duration is always 0 but billsec varies . On 15/08/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR-CSV Processing
Enable mysql loggin of cdr's by installing asterisk-addons and use asterisk-stat http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 On 13/08/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Aug 13, 2007 at 08:49:11AM -0500, Jeremy Mann wrote: Does anyone have any tools to process CDR-CSV files into reports? Throw them into a near-by spreadheet. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
Please stop advertising your forums/services on every single chance u get on users list . On 08/08/07, Al Bochter [EMAIL PROTECTED] wrote: That is why you need to start posting info about the providers at http://www.bochterservices.com/phpbb/ so everyone knows This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. There will be more added by your request Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Stephen Bosch wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000764-2, 08/08/2007 - 8/8/2007 5:31:56 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick sip channel whn two party talking
google for ASTERISK CMD CHANSPY and follow voip-info link in search results . On 08/08/07, satish patel [EMAIL PROTECTED] wrote: Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel -- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool.http://us.rd.yahoo.com/evt=48518/*http://autos.yahoo.com/carfinder/;_ylc=X3oDMTE3NWsyMDd2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDY2FyLWZpbmRlcg--+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk wait for traling digits
This should be configured in phone system instead of asterisk :) . On 08/08/2007, Michael Rice [EMAIL PROTECTED] wrote: This is part f the phones dial plan. Our aastra phones do the same thing. Most phones allow you to configure the dial plan on them. satish patel wrote: i have only one single 16XX dialplan for reached to avaya system then why i have to wait for more digit satish patel */Don Pobanz [EMAIL PROTECTED]/* wrote: satish patel said I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan This part of the dial plan looks like it should dial without the wait. Could there be another part of your dial plan that starts with '16'? If not have you reloaded extenions.conf either by restarting asterisk or doing an 'extensions reload'? Don Pobanz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Pinpoint customers http://portal.mxlogic.com/redir/?atTQSjhOUqenT3qtXTvhvp7ndw0SWt53ySAWRVvfcPeoujvLw1g0tfSdyqKNa_ek2f5J9RHO-r5rablxiIvgF-NIj5j9EVU8AGD1cojjjsqIGIs1Z9RGRqpAUgmy30RGxM7qECsd3rh0V-VK_nLt6WtQXTdTdXivNBgGnrFYq5O5mUm-wafBitegAhASHOVJNdwQsCQknD7TAm1P1JZAS2_id41FrSA_zaxkKTjUQdbFEwSA_zaxkQg6dBcQgeRyq89NQ-k29EwgAhBexKvxYYmfSk3q9J4SDtBxBwQszDC3vZCceKlB who are looking for what you sell. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Rice Systems Administrator Office: 210-366-2500 Ext. : 231 Direct: 210-293-6231 McClelland and Hine, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR/MySQL basic config
sock=/tmp/mysql.sock Is this path for socket correct ? In some distro it is /var/lib/mysql/mysql.sock . Type locate mysql.sock in shell . Also remove uncomment port=3306 if using socket to connect . On 07/08/07, Alessandro Russo [EMAIL PROTECTED] wrote: Hi, try to login as asteriskcdruser to mysql # mysql -u asteriskcdruser -p Enter password: password Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 12 Server version: 5.0.32-Debian_7etch1-log Debian etch distribution Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql Can you login with asteriskcdruser? If you cannot login there are some problems with privileges or...I don't know :( On 8/7/07, Adrian Marsh [EMAIL PROTECTED] wrote: Hi Alessandro, Thanks for that.. I'm pretty sure about the user. I used Webmin to confirm the user configs, but I ran your commands anyway: mysql use mysql ; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select Host from user where User = 'asteriskcdruser' ; +---+ | Host | +---+ | localhost | +---+ 1 row in set (0.00 sec) mysql grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 'asteriskcdruser'; Query OK, 0 rows affected (0.00 sec) But I still get the failure: [Aug 7 15:14:10] ERROR[29103]: cdr_addon_mysql.c:436 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. cdr_addon_mysql.so = (MySQL CDR Backend) [Aug 7 15:14:10] ERROR[29103]: res_config_mysql.c:627 mysql_reconnect: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. [Aug 7 15:14:10] WARNING[29103]: res_config_mysql.c:474 load_module: MySQL RealTime: Couldn't establish connection. Check debug. [Aug 7 15:14:10] NOTICE[29103]: config.c:1171 ast_config_engine_register: Registered Config Engine mysql MySQL RealTime driver loaded. res_config_mysql.so = (MySQL RealTime Configuration Driver) This box also das Cacti installed on it, which makes use of the MySql server as well (and all is ok there). Adrian Marsh -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Alessandro Russo *Sent:* 07 August 2007 14:13 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CDR/MySQL basic config Hi, first step is correct Hmm.. This is what I get: [EMAIL PROTECTED] ~]# mysql -u root -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 187143 to server version: 4.1.20 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. You make an errore here : mysql use asteriskcdrdb users' information are stored in mysql db mysql use mysql; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql mysql select Host from user where User = 'asteriskcdruser' ; +---+ | Host | +---+ | localhost | +---+ 1 row in set (0.00 sec) mysql Are you sure that user 'asteriskcdruser' has the privileges to insert record in DB asteriskcdrdb? If not...allow 'asteriskcdruser' to insert record ^_^ mysql grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 'asteriskcdruser'; mysql exit Reload asterisk and try On 8/7/07, *Adrian Marsh* [EMAIL PROTECTED] wrote: Hmm.. This is what I get: [EMAIL PROTECTED] ~]# mysql -u root -p Enter password: Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 187143 to server version: 4.1.20 Type 'help;' or '\h' for help. Type '\c' to clear the buffer. mysql use asteriskcdrdb ; Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Database changed mysql select Host from user where User = 'asteriskcdruser' ; ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist mysql Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Forrest Beck Sent: 07 August 2007 02:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR/MySQL basic config Adrian, What host/ip did you specify when you created the user? # mysql --user=root --password #mysql use mysql; #mysql select Host from user
Re: [asterisk-users] Use of context=... in [default] section of sip.conf
When you make calls then context=xxx of the peer you are using ( your extension ) will matter , the context=yyy line of your trunk wont matter . If you dont specify a context= for a peer then it is considered to be in [default] context . On 07/08/07, Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-08-07 at 11:18 -0400, Filipe Brandenburger wrote: If I have [myprovider] section with context=something. When I do an outgoing call by using Dial(SIP/myprovider/464646), does context=... affect anything? As I understand it, it only affects incoming calls, but I might be wrong. That's correct. The context is only there to tell Asterisk where in the dialplan to send *incoming* calls. Another thing, the setting of context=... on [default] section will affect all [provider] sections without context=..., right? What if I don't specify any context on [default], what would be the default context? My guess would be the [default] context, but I could be wrong. What if there's no context or an invalid context on a section, what would happen to incoming calls that match that section? The calls would most likely be rejected by Asterisk. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax2 registration being rejected
Yes, since IAX2 only uses one port, this is correct. Another thing to keep in mind is to set a low qualify value in Asterisk since some routers will tear down the connection pretty quickly. The qualify acts as a keep-alive and prevents the router from closing the port and losing the map. Thanks, Steve But if you set timeout lower than actual latency to peer .. it will result in asterisk not sending any calls to peer at all so keeping it too low will create more problem .. however peer will be able to make outgoing calls . I think asterisk doesnt rely on qualify= parameter to keep connection open . Main purpose of qualify option is to make sure peer is not lagged then specified timeout period else call quality will be pathetic .. qualify=200 seems ok . Btw i have never seen a device losing registration when qualify value is set huge ( i keep qualify = 2000 for a very dirty connection sometimes :D so that asterisk will show latency when i do sip show peers and iax2 show peers in cli ) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] partial ChanSpy
Chanspy() app allows spying live channel but you will get 2 way voice in it . I dont think any other app allows to spy on one side of call . On 03/08/07, nik600 [EMAIL PROTECTED] wrote: Hi is it possible to spy (not record, spy) partially on a channel? for exaple, i'd like to listen only the input or the output voice. is it possible? thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1and1 dedicated servers have been down for a few hours .
Is this a web hosting forum or mailing list ? On 31/07/07, Asterisk guy [EMAIL PROTECTED] wrote: 1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly MeetMe() question.
Do you have proper version of zaptel installed corresponding to your asterisk version ? On 31/07/07, Knud Müller [EMAIL PROTECTED] wrote: Alex Balashov schrieb: On Mon, 30 Jul 2007, Knud Müller wrote: what does your modules directory contain? Can you find a file /usr/lib/asterisk/modules/app_meetme.so after make install? No. I know it needs to be compiled, but it is not being compiled no matter what I seem to do in the way of arguments to ./configure, installations of zaptel, etc. Better have a loot at the apps directory, there is a Makefile that lists all apps to be compiled. app_meetme depends on a flag called WITHOUT_ZAPTEL. I have not tried to use meetme without zaptel, but its worth a try add meetme explicitly. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Knud A. Müller Geschäftsführer Tel.: 040/398053-11 Fax: 040/398053-29 e-Mail: [EMAIL PROTECTED] portrix.net GmbH Stresemannstr. 375 22761 Hamburg HRB 79850 (Amtsgericht Hamburg) Geschäftsführer: Knud Alex Müller, Henning Voss, Niclas Schroeder http://www.portrix.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking a device to a codec
in ur sip.conf under the device definition you can set it for example device name is asterisk is pap2 [pap2] username=pap2 secret=blabla type=friend disallow=all allow=g729 Then asterisk will only use g729 for incoming as well as outgoing calls on this device . On 27/07/07, Matt [EMAIL PROTECTED] wrote: Right.. what I'm asking is: If I set my PAP2T to use G723 or G729 outgoing calls from that device go in that format. However, incoming calls to the device from asterisk are running at G711u. The PBX will access any format G711u, G723, G729 or GSM. What do I need to do to make asterisk use the same codec back to the ATA as it is using to the PBX? On 7/27/07, dave cantera [EMAIL PROTECTED] wrote: baji, mhoppes, remember, if you have Only the g729 codec allowed or if this is the only allow= entry in the sip.conf file, callers requesting any other codec will be rejected daveC Baji Panchumarti wrote: On 7/27/07, Matt [EMAIL PROTECTED] wrote: Can someone comfirm my logic here? If I want a phone to use G729 I can set it to use G729... do I also need to set it in Asterisk? I'm thinking no... as long as asterisk WILL do G729... if that's all the device accepts it should go to that codec, yes? (based on my understanding, take it for what it is worth) if allow=all or allow=g729 is in your asterisk configuration (sip.conf / iax.conf ) then asterisk will stream packets in g729 (assuming you have any licesnses needed in place). -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Max Channels Setup
http://www.voip-info.org/wiki-Asterisk+config+sip.conf * call-limithttp://www.voip-info.org/wiki/edit.php?page=Asterisk+sip+call-limit * = number : Number of simultaneous calls through this user/peer On 27/07/07, Nicholas Blasgen [EMAIL PROTECTED] wrote: I'm running Asterisk without FreePBX or any of the other managers. I'm trying to figure out how to set the maximum number of channels allowed on a single line? I'd just rather not have Asterisk try the line when I know I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this case). Is there a configuration option I can't find that sets the maximum number of connections a SIP channel can handle at a given moment? I expect the line to be something simple, but I can't find it detailed on the Wiki. -- /Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best softphone work with Asterisk
Idefisk is now renamed to zoiper . http://www.zoiper.com/ :) On 26/07/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad: Hi BaharatSamaria; Thanks for your kindly email. Are (Xlite and phoner) IAX or SIP? From where I can download them (Xlite and phoner)? I googled for xlite. One of the first matches was a wiki page on voip-info.org, which in turn linked me to the X-Lite manufacturer's homepage. quote CounterPath's X-Lite 3.0 is the market's leading free SIP based softphone available for download. /quote. The first link in the google search list for phoner immediately led me to the phoner homepage, quote - VoIP support for SIP connections Phoner is freeware, so this program can be used and distributed without any restrictions. Distribution has to be free of charge. /quote I think you will have no trouble to find the URIs yourself, probably within about 30 seconds. In doubt you might consult http://www.googleguide.com/ to learn about google. Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lines Not being Hung UP Major
Btw are the phones behind NAT ?? Also you can try some softphone and make sure that this problem is caused by snom phones or some other factors .. On 25/07/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I thought it was the fios service but now I realize it's the snom 360! It doesn't hang up random outgoing calls. It seems like it only happens on outbound calls from phones that have been updated to 6.5.12 or 6.5.10. It didn't happen before, but I don't remember what version firmware it was before, maybe 6.2.3 or so. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron Arts Sent: Monday, July 16, 2007 5:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lines Not being Hung UP Major Do your SNOM phones sometimes use answer-after:0, and do they have keyboard LEDs subscribed to their own extensions? Do those people hangup calls by puttig down the handset instead of pressing the X key? We are seeing hanging channels in this particular case. Ron Michael J. Liberatore wrote: Hi all, i am having a major asterisk problem. I think it started around 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360. basically we start getting busy signals, all our 4 line hunt group is busy, i then check the channels and there are open calls that were hung up long ago. i thought it was a zap problem but then i saw the same problem with iax2 calls. its becoming a huge issue because if i dont reboot asterisk several times a day, all our lines get filled up with dead calls. I am now running 1.2.21.1 asterisk with the same problem. Please help. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best softphone work with Asterisk
Idefisk/zoiper softphone is for IAX2 and it works fine almost everytime . However there is more variety in sip softphones . I think zoiper is much better than other iax2 softphones . On 25/07/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; Thanks for all replies :) - But that means, softphone in Asterisk is not that good, I see all complains. Any advise? Please Mr. Time Bandit: What do u mean by my IAX2? Is it your code or what? Also Mr. Rayan: I am noticing that you are advising for SIP, what about IAX? Nothing suitable? If this is the case, then where is the main advantage of IAX protocol as specifically the IAX softphone does not work fine? Any help? Regards Bilal I've had decent luck with PhonerLite, connecting via SIP. The interface is not the best, but I've been able to connect reliably and make calls. -Ryan bilal ghayyad wrote: Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP IP Trunk, between Asterisk and Softswitch
In your case it will send calls without registering to softswitch . Btw what does your softswitch expects from asterisk ? like is it configured to authenticate by username alone , user/pass or ip address ?? People here can help you better if you post that info . On 24/07/07, bilal ghayyad [EMAIL PROTECTED] wrote: Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk request to register on the softswitch or it can send directly without registeration? (Note: the trunk is SIP). Please check the below configuration and advise me if it is correct: [aloonet] type=peer qualify=yes host=193.111.196.240 ; IP Address of the softswitch canreinvite=yes context=outbound disallow=all allow=g723 nat=no Is it OK? Will it register on my softswitch or will send call directly without registeration on it? Regards Bilal Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk is not sip proxy
Asterisk is not a sip proxy but it *can* partly act as a sip proxy if reinvites are enabled ( canreinvite=yes in sip.conf ) only then asterisk connects 2 end points directly and does signalling between them . Asterisk is a PBX now suppose u need to record all calls ..do conferencing stuff then rtp stream need to pass from asterisk ( openser cant do this bcoz it just connects 2 endpoints and only does signalling ) .. If you do canreinvite=yes in sip.conf for both peers then asterisk does only signalling ( also dial command should not have transfer parameters tT .. ) . If both peers are behind NAT then asterisk reinvites may not work properly . On 23/07/07, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Edgar; I am little bit confused, do u mean that asterisk does not work in that way: RTP (media) to be from the sournce to the destination directly while signaling to be via asterisk? So, what he parameter canreinvite is doing? Regards, ITS Ip Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Hello Asteriskers, I'm confused about why Asterisk is not a SIP proxy and why exactly this can affect the performance of a large Asterisk system. I know that Asterisk acts as a useragent endpoint, but my doubt is why exactly Asterisk could overload the call flow if the RTP voice stream goes from the caller to the called party. Does someone know how many calls or pencentaje that could handle a SER or OpenSER in comparison with Asterisk? Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
Get portsip ( www.portsip.com ) its realtively easy to configure ( just push in user/password and server name at startup ) .. there might be NAT issue so make sure you have nat=yes in ur asterisk's sip.conf for the peer definition . If it still doesnt work then you need to find a iax phone like zoiper ( http://www.zoiper.com/ previously idefisk ). On 21/07/07, WipeOut [EMAIL PROTECTED] wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which features are lost when canreinvite is turned on ?
If you manage to get everything working with canreinvite=yes ( i suppose u figure out nat issues ) then you cant play music on hold , can't record calls , and can't do most of pbx stuff asterisk is capable of .. but dont worry asterisk doesnt disable all this features if canreinvite=on .. like if you have call recording enabled in configuration and also have canreinvite=yes then asterisk wont send reinvite's and media stream will pass thorugh asterisk . For most of pbx canreinvite should be kept off unless you have latency issues , or you are just connecting 2 pbx systems and doing something like billing in between and not touching media stream . On 09/07/07, Olivier [EMAIL PROTECTED] wrote: You mean I'm heading to NAT issues ? And what about Record-Route options ? Will it really help to be notified of call endings ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk console filtering and logging
This feature would be really great but i dont think asterisk supports it . It either shows dialplan execution of all extensions when verbosity is increased or of none when set to 0 . You can set verbose 0 and sip debug a single peer but you cant enable dialplan execution viewing for single extension/peer ( please correct me if i am wrong ). Regards, Jaswinder Singh. On 05/07/07, Eugene Prokopiev [EMAIL PROTECTED] wrote: Hi, Is it possible to filter messages on asterisk console, which was started with -, to see messages only for one extensions? By default there are all messages for any extensions displayed so dialplan debuging is very difficult. Is it possible to log such console messages: ... -- Executing Set(SIP/10.0.0.1-0061f5d0, CDR(userfield)=2422718) -- Executing Dial(SIP/10.0.0.1-0061f5d0, SIP/708,25,tT) ... to file. I can't find any suitable option in logger.conf -- Thanks, Eugene Prokopiev ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade Asterisk
Yes just download new version of asterisk,zaptel,libpri . make install for all 3 ( first libpri , then zaptel, then asterisk ) . It is recommended to stop asterisk b4r doing make install of new version . Do not do make samples or it will overwrite you config's . After installing newer zaptel do rmmod ztdummy zaptel zttranscode then modprobe 3 of them ( or a restart of server will do ) . Now just start asterisk again and it will read all the prior config's you made as they are in /etc/asterisk . It's that easy :) . or just do make install for all 3 packages and restart server once ( it will load new kernel modules after restart automatically and you dont need to do that rmmod and modprobe stuff ) . On 04/07/07, Christian Victor [EMAIL PROTECTED] wrote: Hi! Just ashort question - obviously I am too stupid too find the answer on the net. :-) I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have to do? Just install it over the existing version? Do I need to backup the configuration? Will I need to reconfigure the source or will the new version import my old settings? Will I need to update Zaptel and Libpri too? Argh - I installed like 50 asterisk systems but this one is the first production machine with issues so heavy that I have to upgrade it. Please point me to a update/upgrade howto etc. if available on the net. Thanks a ton Christian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple CDRs w/Asterisk/OpenSER.
Asterisk is poor with codec negotiation . It does not check if it can avoid transcoding by forcing codec available to both sides .. instead it will read it's config file and will select first allowed codec that is also available on other device on each leg of call and happily transcode between them .There was a patch on digium submitted by someone for asterisk 1.2.12 or so but it isnt updated from long time . I am sure guys at digium are aware about it and working on it . It's not a bug since asterisk is not a sip proxy and tries to keep media path through it to offer its pbx features but it would be a great feature nonetheless if implemented . On 05/07/07, Alex Balashov [EMAIL PROTECTED] wrote: Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based proxy / call routing setup? I need to get simple CDRs; not for detailed settlement/rating, but just for reconciliation with an ultimate TDM carrier just to make sure we only get billed for what we're actually using. I'd use the often-heralded approach of dumping a call from OpenSER into Asterisk and having it bounce right back out toward the proxy by way of REINVITEs. I don't want the media running through Asterisk or Asterisk being a limiting factor in that regard. The problem is I don't have native G.729 support - we have no need for it because neither the customer's network elements nor ours lack an implementation of their own they can negotiate on just fine. But unfortunately Asterisk insists on natively homogenising the SDP from both sides even if it subsequently removes itself from the media path! So, I end up with situations where on the one side, I get, say: Customer MGW -- OpenSER -- Asterisk - sends call as G.729. Asterisk -- OpenSER -- Our MGW - our MGW prefers G.711a. Now, if customer MGW - Our MGW were talking directly, as they do when the deal is brokered through the OpenSER proxy, they would simply negotiate upon what they agree. But for some reason with Asterisk this does not seem to be working as advertised; we get lots of failed calls if we pass them through Asterisk because one leg is one codec and the other is another. I am not sure how it arrives at that conclusion despite the overlap of shared codecs (G.729 on both sides, I would expect it to pass thru licence-free), and to be honest, I don't particularly care if it's a bug or a feature, I just need it not to introduce codec issues if I use it as a billing target. Any help or insight would be greatly appreciated. Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade Asterisk
Yes that is write order . libpri then zaptel then asterisk . Remember that zaptel compilation is not required if you are using asterisk for voip only environment .But it's always good to install it before asterisk if you want to use conferencing abilities of asterisk . Regards, Jaswinder Singh On 05/07/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Hi, Try first installing latest release of libpri, then zaptel Try install asterisk after then. ope you will be able to compile it without any probs. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about dnsmgr
Are you sure calls were dropped with change in IP ?? I think it should let current calls run and use new IP for new connections . However if destination serv drops calls then it's a different story . On 03/07/07, Henry J. Cobb [EMAIL PROTECTED] wrote: Asterisk 1.4.5 full log: [Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups. [Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net' changed from 64.2.142.17 to 64.2.142.29 [Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed Jitterbuf max 600 timeslots And the calls are dropped. I fixed this by turning off enable in dnsmgr.conf My question is: Do you attempt to move existing IAX connections when you see a DNS change or do you leave the existing connections the fnord alone on their current IP addresses and simply use the DNS change for new connections? -HJC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google acquires Grand Central
Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . On 03/07/07, Dean Collins [EMAIL PROTECTED] wrote: Ooops did Google just become a carrier :) http://googleblog.blogspot.com/2007/07/all-aboard.html I hear stocks crumbling worldwide as I type. Cheers, Dean ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] v1.4.x ready yet?
I jumped into asterisk 1.4 and its pretty stable .. i never got a core dump but it did halt while reloading a few times . I am back on asterisk 1.2 now but i think asterisk 1.4 is stable . On 29/06/07, Bruce Reeves [EMAIL PROTECTED] wrote: While I have not jumped all my systems to 1.4, there were some that I have moved to 1.4 and I have found it to be as stable as 1.2 was on those machines.One of the systems is a 10 user office with Sangoma cards and another is a 70+ user pure voip system. In both cases I have warning messages about my dialplan usage of realtime and the fact that it will be depreciated in the next release, but everything works as it should and the upgrades.txt guided me through the changes to my dialplan. Hope that helps. On 6/29/07, shadowym [EMAIL PROTECTED] wrote: Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + hinting presence + macro
It was due to changes in cdr in asterisk 1.4.5 previous version does not do
it .there is a fix on bugs.digium.com or you can wait till next release or
use asterisk 1.4.4
On 28/06/07, Rob Schall [EMAIL PROTECTED] wrote:
I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set up notify status.
On voip-info, they say do something like...
,hint,SIP/
,1,Dial(SIP/)
blah blah blah
This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}
this adds a s hint which obviously doesn't work, instead of a hint for
as it should.
Any ideas?
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Re: [asterisk-users] Asterisk + hinting presence + macro
Sorry i didnt read your mail properly . I thought your problem is with
cdr's. Here's link to cdr problem :)
http://lists.digium.com/pipermail/asterisk-dev/2007-June/028085.html
see the next message for patch .
On 29/06/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Rob Schall wrote:
Eric ManxPower Wieling wrote:
Rob Schall wrote:
I currently have about 50 polycom 501 phones on my asterisk setup.
The dialplan is set to work with mysql (realtime), and all of the
extensions for the phones route through the same macro (stdexten).
This all works fine until I tried to set up notify status.
On voip-info, they say do something like...
,hint,SIP/
,1,Dial(SIP/)
blah blah blah
This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}
this adds a s hint which obviously doesn't work, instead of a hint
for as it should.
Yes. Put in the correct hint. There is no reason that
,hint,SIP/ would not work in a macro.
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So, if I understand you correctly, my macro would look something vaguely
like...
[macro-stdexten]
${ARG1},hint,SIP/${ARG1}
s,1,Dial(${ARG1})?
This will work? My understand was that by going into a macro, you were
going to be using the s extension. I'm not sure how that hint would
get called if its not inside the s extension.
I have no idea, but as I understand it, Hints are separate from
extensions.
I guess you could do something like:
[macro-stdexten]
exten = s,1,Goto(${MACRO_EXTEN},1)
exten = _,hint,SIP/${ARG1}
exten = _,1,Dial(${ARG1})
I do this sort of thing in many of my macros that Dial somewhere. I
seem to remember something about hints not working for pattern matching.
or working weirdly.
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Re: [asterisk-users] CDR Records s as dst
This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
it was intended to be that way .
On 25/06/07, Troy - Purple Oranges [EMAIL PROTECTED] wrote:
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of s as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work around so that I can record the actual dialed number?
[macro-dialout]
exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = s,n,NoOp(CID_NAME : ${CID_NAME})
exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = s,n,NoOp(CID_CLIR : ${CID_CLIR})
exten = s,n,NoOp(TRUNK : ${TRUNK})
exten = s,n,Set(CALLERID(name)=${CID_NAME})
exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
exten =
s,n,Set(PRESENTATION=${IF($[${CID_CLIR}=1]?prohib_not_screened:allowed_not_screened)})
exten = s,n,SetCallerPres(${PRESENTATION})
exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW) ;Ring the interface
exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1) ;Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-BUSY,1,Playtones(busy)
exten = s-CONGESTION,1,Playtones(congestion)
exten = _s-.,1,Goto(s-CONGESTION,1) ;Treat anything else as no answer
--
Regards,
Troy Kelly
Director
Purple Oranges Pty Ltd
http://purpleoranges.com/
--
Brisbane (07) 3018 2840
Fax (07) 3105 5987
Disclaimer - This email and any files transmitted with it are
confidential and contain privileged or copyright information. You must
not present this message to another party without gaining permission
from the sender. If you are not the intended recipient you must not
copy, distribute or use this email or the information contained in it
for any purpose other than to notify us.
Any views expressed in this message are those of the individual
sender, except where the sender specifically states them to be the
views of Purple Oranges Pty Ltd.
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Re: [asterisk-users] Binding to multiple addresses
You can use bindaddr=0.0.0.0 to bind to all interfaces in sip.conf and iax.conf . On 23/06/07, Jordan Novak [EMAIL PROTECTED] wrote: I have a simliar problem as the port binding question. I have a four port parelell processing NIC, I would like to team them together. Can I do this in asterisk if they are not actually teamed in hardware. I would be binding to several addresses simultaniously. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of ChanSpy
exten =*76,1,Answer
exten = *76,2,Chanspy(|qb) ; q for quiet and b for only bridged calls
exten = *76,3,Hangup
Now you can spy on any call ,. All you need to do is press * again and again
to change calls . Like if 3 calls are going then you can switch between
calls by pressing * and # increases or decreases volume
This will spy on only sip calls:
.exten = *76,2,Chanspy(SIP/|qb)
for iax2:
.exten = *76,2,Chanspy(IAX2/|qb)
for a certain extension ( eg: sip extension 4455)
.exten = *76,2,Chanspy(SIP/4455|qb)
Hope this helps
On 24/06/07, Oscar Carriles [EMAIL PROTECTED] wrote:
Maybe this helps
; spy on agent
exten = *7792,1,Playback(agent-newlocation)
exten = *7792,2,Read(EXT)
exten = *7792,3,Chanspy(Agent/${EXT}|q)
exten = *7792,4,Hangup
; spy on sip
exten = *7797,1,Playback(agent-newlocation)
exten = *7797,2,Read(EXT)
exten = *7797,3,Chanspy(SIP/${EXT}|q)
exten = *7797,4,Hangup
; spy on everybody
exten = _**779.,1,Chanspy(${EXTEN:5}|q)
exten = _**779.,2,Hangup
-Mensaje original-
*De:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *En nombre de *Carlos Garcia
Mujica
*Enviado el:* Jueves, 21 de Junio de 2007 04:17 p.m.
*Para:* asterisk-users@lists.digium.com
*Asunto:* [asterisk-users] Use of ChanSpy
How can I use the Asterisk command ChanSpy If I need to spy on a call?
I already added the function to the extensions.conf, and I get the beeps,
but then what do I do??? I don't understand the use of this function.
Best Regards
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.9.6/865 - Release Date: 24/06/2007
08:33 a.m.
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.9.6/865 - Release Date: 24/06/2007
08:33 a.m.
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Re: [asterisk-users] SIP Peering--call terminated prematurely
Please do not post same thing again and again . It wont help you get better replies , Post you asterisk cli output while call is in progress and when it disconnects prematurely . On 18/06/07, Don Kelly [EMAIL PROTECTED] wrote: I am attempting to establish SIP peering between Asterisk and an AltiGen soft PBX. This is my first experience with SIP peering. I can successfully make both inbound and outbound calls to/from a softphone on the AltiGen system (network access is provided by a PRI on the Asterisk system), but they are disconnected unexpectedly. The attachment is a redirect of the Asterisk CLI during a call that is disconnected prematurely. Here's what's in SIP.conf: [altigen] type=friend username=altigen secret=coolbeans host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.2.150/255.255.255.255 qualify=yes disallow=all allow=ulaw context=altigen-inbound dtmfmode=rfc2833 The machines are a couple feet apart on a LAN through a 100MB switch. I'd appreciate any help. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ --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] Qualify renders all SIP peers unreachable
What does sip show peers output ? Also set a timeout in millisec like qualify=200 instead of qualify=yes On 14/06/07, randulo [EMAIL PROTECTED] wrote: I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT. All SIP peers are working properly to place or receive calls. Any SIP peer or friend whether NATted or not will become UNREACHABLE if qualify=yes. I have identical peers on the other asterisk 1.2.16 production server. In fact, two of the phones (linksys 941 and Polycom ip500) are using one line for each asterisk. The 1.2 one works normally, the 1.4 does not. The sip confgs from sip show settings are identical on the two servers. The sip.conf peer entries were moved over exactly. Ports 5060 to 5065 are forwarded to the asterisk server. Looking at sip debug, I notice a few differences: REGISTER from phone: Authorization: Digest username=Poly, realm=asterisk,... does not show on the 1.4 server. Trying (sent by *): Supported: replaces The Via lines are the same (internal ip addresses) on both servers, but there is a Sending to 192.168... on the 1.2 message where there is none on the 1.4. What is supported: replaces ? What config setting generates the Authorization: Digest... message ? ___ --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] No audio after Dial with G option
Remove Answer() and try .
On 12/06/07, Rosalinda Trevino Cadena [EMAIL PROTECTED] wrote:
I'm using the Dial application in the extensions file with the G option
to execute an AGI script after the Dial (I need to track the call status) as
follows:
exten = _X.,3, Dial({DIAL_STRING},,G(_X.^4))
exten = _X.,4, Answer()
exten = _X.,5,AGI,agiScript.php
The problem is that testing between two internal phones (with two ATA) I
loose the audio when I include the G option in the Dial application, while
the audio is restored if I remove the G option, but that way I can't execute
the AGI script wile the call is up.
Any ideas on how to solve the problem or on what the cause might be?
Thanks,
Rosalinda
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Re: [asterisk-users] IAX trunk with dynamic IPs
In your no-ip client set it to update ip every 2 minutes or so . and /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt its 300 ( 5 minutes) On 10/06/07, Ronaldo Z. Afonso [EMAIL PROTECTED] wrote: Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: *set enable=yes in the [general] section of /etc/asterisk/dnsmgr.conf* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX trunk with dynamic IPs
Hello You should use qualify=310 ( any value in millisec ) .. qualify=yes is not proper . I am not sure about how asterisk's dnsmgr manages dns refreshing but maybe someone else can answer that question . On 10/06/07, Ronaldo Z. Afonso [EMAIL PROTECTED] wrote: Hi Jaswinder, That is what I did. The thing now is, when I set enable=yes in /etc/asterisk/dnsmgr, Asterisk stops sending IAX POKE messages to the remote peer (to keep the trunk up). I've searched on the Internet but I couldn't find any documentation about how DNS update manager works for Asterisk. Do you have any? Ronaldo. Jaswinder Singh wrote: In your no-ip client set it to update ip every 2 minutes or so . and /etc/asterisk/dnsmgr.conf set refresh interval as 30-40 by defualt its 300 ( 5 minutes) On 10/06/07, Ronaldo Z. Afonso [EMAIL PROTECTED] wrote: Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: *set enable=yes in the [general] section of /etc/asterisk/dnsmgr.conf* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] g729
Are you trying to record the conversation as well ? On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote: I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies ___ --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] g729
I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
? I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
Yes
This is my extensions.conf entry.
exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729
Are you trying to record the conversation as well ?
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also
set to g729. The call goes through just fine, but these error messages
keep
flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
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Re: [asterisk-users] Voip-info.org
Yep its down for me tooo . On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote: Is anyone else having trouble going into voip-info.org today? ___ --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] g729
Just read somewhere that you can use extension as g729 even in
mixmonitor so it will record g729 stream and later you can convert it
to mp3 or wav using sox . If this fails then try monitor application .
On 06/06/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
? I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
Yes
This is my extensions.conf entry.
exten = _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten = _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten = _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten = _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten = _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten = _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729
Are you trying to record the conversation as well ?
On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also
set to g729. The call goes through just fine, but these error messages
keep
flying by until I disconnect the call.
Any ideas?
ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
Jun 5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
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Re: [asterisk-users] any codec passthru mode
Yes it might be dumb but since asterisk is a pbx and not a sip proxy it has to perform many other functions as well . But i do think that asterisk should act little smart in this case SIP wrote: That just seems really, REALLY dumb for a program of this magnitude. I know this has been patched here and there by one person or another, but does anyone know if any of these patches to make CODEC negotiation actually, you know, negotiate a CODEC will ever make it into the core src? Jaswinder Singh wrote: Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding but if one side prefer gsm and other prefers g729 and the gsm side can also support g729 still asterisk will transcode . Someone posted a patch to this in mantis bug tracking system at digium for 1.2 .. google for it and maybe you can find . On 31/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Does anybody has any documentation on codec negotiation within asterisk? Well im using free g729 codec for testing purposes. i mentioned g729 just as an example. whatever codec is mentioned in user perefernce, asterisk uses ulaw to throw out the call. On 5/30/07, Marco Mouta [EMAIL PROTECTED] wrote: so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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 -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] TCP-UDP SIP proxy?
I think there is a patch for sip over tcp in asterisk but not sure if its stable or not try this http://bugs.digium.com/view.php?id=4903 You can also install openser as sip proxy . it supports sip over tcp . On Wed, 6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? Thanks! __Yehavi: ___ --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] Voip-info.org
Its up and working now . On 06/06/07, Compnet Bobby [EMAIL PROTECTED] wrote: Same in southern cali! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Wednesday, June 06, 2007 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voip-info.org Yep its down for me tooo . On 06/06/07, Ed Nuñez [EMAIL PROTECTED] wrote: Is anyone else having trouble going into voip-info.org today? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062
In sip.conf it should be bindport=5062 On 06/06/07, Crazy Boy [EMAIL PROTECTED] wrote: Hi Friends, I want to use 5062 port for SIP protocol. I made the below modifications in my server to use 5062 port. Polycom phone: port=5062 Trunk settings: port=5062 sip.conf: bindaddr=5062 Extension configuration details: 5062 Our VoIP provider told me that they are allowing the SIP traffic through 5060 to 5064. I observed on my server console that my server is registered with our VoIP provider with 5062 port. But, I am unable to make outgoing calls. Do I need to modify any other settings in Asterisk? Look forward to your response. Thank you. Regards, Chandra. Need a vacation? Get great deals to amazing places on Yahoo! Travel. ___ --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] yum om centos
independently install each rpm via rpm command :-/ On 04/06/07, Khaled Chehab [EMAIL PROTECTED] wrote: I have 2 servers, one connected to internet and the other is on a private lan have no access to internet. On the first server I update the kernel by yum update And installed asterisk prerequisite module yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel I zipped /var/cache/yum from the first server and extract it on the second server at the same directory. On the second server I tried to update using yum update but the yum update failed. How can I do that with out connecting the second server to internet . Khaled Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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] any codec passthru mode
Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding but if one side prefer gsm and other prefers g729 and the gsm side can also support g729 still asterisk will transcode . Someone posted a patch to this in mantis bug tracking system at digium for 1.2 .. google for it and maybe you can find . On 31/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Does anybody has any documentation on codec negotiation within asterisk? Well im using free g729 codec for testing purposes. i mentioned g729 just as an example. whatever codec is mentioned in user perefernce, asterisk uses ulaw to throw out the call. On 5/30/07, Marco Mouta [EMAIL PROTECTED] wrote: so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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 -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --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] ringback detection
It just might be that your carrier is not sending ring . You can use 'r' in asterisk dial command in extensions.conf to generate ring from asterisk . On 31/05/07, dima [EMAIL PROTECTED] wrote: Hello, everyone. Could anyone explain me how does ringback detection works in asterisk. Sometimes, when making a call, my asterisk box doesn't detect a ringback and I just hear silence until the other party picks up the phone. I've checked the SIP messages and they are ok (I'm getting 183 session in progress), so I guess I should be debugging the RTP packets. From then on I'm stuck. Does anyone know what type of packets I should be looking for? ___ --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] Net2Phone Multiple SIP Trunk Not Working
You just have a 1 call limit on your account on net2phone side . Making 10 trunk wont let you make 10 account its restriction on your account not ip . Just change your provider . On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote: Hi, Any help regarding Net2Phone poblem? BR On 6/1/07, Andrew Furey [EMAIL PROTECTED] wrote: On 01/06/07, Salah Eddine ELMRABET [EMAIL PROTECTED] wrote: I'm sorry that's because I didn't get a visibility of ny post, I though that was a network problem (as I cannot see my post on the mailing list) You never do with mailing lists on Gmail, I presume it hides it based on the message ID (since you already have a copy). Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!
Well freepbx runs fine with asterisk 1.4.4 ( atleast for me ) i think some changes was introduced in 1.4 ( 1.4.4 ?) for some backward compatibility... like show channels now work in 1.4.4 instead of core show channels but it gives a notice that 'show channels' is deprecated bla bla .Freepbx works completely fine with asterisk 1.4 for me . On 31/05/07, shadowym [EMAIL PROTECTED] wrote: If anything this should motivate the FreePBX developers a bit more. -Original Message- From: Jared Smith [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 30, 2007 12:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO! On 5/30/07, BSumrall [EMAIL PROTECTED] wrote: AMP does not support 1.4 and will not until AMP 2.3 is released! I'm sorry to hear you think our decision (I say our, as I was at the Asterisk Developers' Conference where the decision was made) will kill the AMP project. Personally, I don't think the situation is as dire as you say. I'm quite sure the AMP developers will step up to the plate and support Asterisk 1.4 in due time. When that will be I can't say, as I'm not active in the AMP community. I can't image it would take that long to move over to Asterisk 1.4, as the dialplan changes aren't *that* extensive between 1.2 and 1.4. (Obviously any code that ties into the internal C APIs of Asterisk will take longer to port.) Bet you guys didn't think about that one! Actually, we did. As a matter of fact, I was *very* vocal at the conference in stating that we needed to give users, integrators, and projects like AMP a substantial warning before putting Asterisk 1.2 in security maintenance mode, as they need time to react. At the same time, I don't think anyone should expect the Asterisk developers to base all their decisions completely on the timetables of outside projects (like AMP). There is a plethora of projects and programs out there that tie into Asterisk, and if we as developers waited for every single one to move over to Asterisk 1.4, we'd never accomplish anything. There's simply a finite set of resources (developers and bug marshalls in this case), and a decision had to be made on how best to use those resources. Personally, I think it would be great if there were more communication between the outside projects and the Asterisk developers, so that there isn't so much animosity when decisions like this are made. In short, the decision is probably going to cause some short-term discomfort for some people, but I truly believe it's a good decision for the long-term health and sanity of the Asterisk developers and Asterisk community in general. No, we're not trying to kill off AMP or any other outside project -- we're trying to make Asterisk (and by extension, anything that uses or adds on to Asterisk) as great as possible. -Jared ___ --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] multiple host= in sip.conf
I dont think asterisk supports this . You can have host=dynamic and he can send calls from different servers . Problem will arise when you need to call him ( if registrations are enabled then latest registration will be getting call from you or you can directly send calls to his ip . ) On 30/05/07, Yusuf [EMAIL PROTECTED] wrote: Hi, I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to call my server and place calls. However, he has multiple IP's that he comes from, and since I authenticate him of his IP, I did this, and it works. [vz1] context=outbound type=friend host=x.x.x.x disallow=all allow=alaw canreinvite=no [vz2] context=outbound type=friend host=y.y.y.y disallow=all allow=alaw canreinvite=no [vz3] context=outbound type=friend host=.z.z.z.z disallow=all allow=alaw canreinvite=no However, is there anyway I can have just one account for him, with mult host= statements, so I can authenticate him based on his IP in just one place? -- thanks, Yusuf ___ --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] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!
Can you post some output from asterisk cli output while you make call ?
On 30/05/07, BSumrall [EMAIL PROTECTED] wrote:
after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
I am down to this.
extensions.conf
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
[default]
exten = 8005181896,1,Dial,(IAX2/UXMC)
exten = s,1,Answer()
(I tried)
exten = _1XX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr)
(as well)
iax.conf
[general]
port=4569
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
register = :[EMAIL PROTECTED]
[teliax]
context=default
type=friend
host=voip-co3.teliax.com
auth=md5
user=
secret=x
disallow=all
allow=ulaw
allow=alaw
allow=gsm
sip.conf
[UXMC]
user=xxx
context=internal
type=friend
qualify=yes
nat=no
secret=
canreinvite=no
host=dynamic
nat=no
If I put back previous config, I can call into the 1800 number and here
that silly chick heckle me from my server!
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Re: [asterisk-users] False ring problem
Do you have 'r' in ur dial command ? 'r' makes asterisk produce ring . On 30/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R and r options in Dial application but they dont work. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
I think its a fair decision . 1.2 is very stable and they are not closing it all together , security issues will still be fixed . They need to concentrate more on 1.4 to make it bugfree . On 29/05/07, Stephen Bosch [EMAIL PROTECTED] wrote: John covici wrote: I have an install using Rhino cards -- I sure hope they get their act together by then. They have no choice now, do they? Nothing focuses the attention like a deadline. -Stephen- ___ --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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
Well i guess you just need a good look on logs for why and when you are getting core dumps . We are having few servers running .1.2.18 and it has turned out to be most stable in whole 1.2 branch ( had some issues with 1.2.13 and 14 ) . Except that for some users 1.2.18 is NOT stable. I've had to roll back to 1.2.15 on my production servers in order to prevent core dumps at least once per day. No, I am not willing to turn my production servers into testing servers to solve this. Doing so would make me a former consultant for these customers. ___ --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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
What you say might be true for small business or home pbx systems . But if you have a production server handling sip/iax trunks over internet then you need to upgrade to avoid security related bugs and exploits that are released . You seem to miss the idea here. You work with a version that supports your feature needs and find the sub-version that provides the most stability for your deployments. Lets face it these boxes should go in and run for weeks, months or even years without much intervention (assuming the mission of the box does not change). I'm running a 1.2.7.something (i think) that has been running almost nonstop since installing. Very reliable and stable for my needs. Compared to a Merlin or Nortel or any other system out that I feel I have a much better product. Could I benefit from a newer sub-version? Maybe. Will I upgrade the box in it current roll? No. Unless the application I use the box for has a major change (or the hardware dies) I'll just let it keep on running as it is. For my future deploys I am working closer with 1.4. The reason is clear. 1.4 is the future of asterisk. When 1.6 or 2.0 comes out I'll investigate into migrating in that direction at that time because that will become the future of asterisk. ___ --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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
Well if you are out of luck with asterisk .. How about its fork callweaver ? I am highly awaiting its stable release to see if it holds upto what its wiki says . On 30/05/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote: Michael Collins wrote: I think its a fair decision . 1.2 is very stable and they are not closing it all together , security issues will still be fixed . They need to concentrate more on 1.4 to make it bugfree . Fair indeed. I would guess that a completely stable 1.2 w/ security maintenance is acceptable to the majority of users. Those folks still using 1.0.x certainly aren't clamoring for new features! The great many Except that for some users 1.2.18 is NOT stable. I've had to roll back to 1.2.15 on my production servers in order to prevent core dumps at least once per day. No, I am not willing to turn my production servers into testing servers to solve this. Doing so would make me a former consultant for these customers. So basically what you're saying is that some efforts should be concentrated on 1.2 as well. So let's start with your specific problems. Are there open bugs for them? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] channel_find_locked: Avoided deadlock
Is it over iax and there are lot of outgoing channels ? If yes then you are not the only person having this .. On 30/05/07, ram [EMAIL PROTECTED] wrote: Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram ___ --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] Meet me
change conf = 222
to conf = 222
( remove | )
I had same problem as freepbx always put | removing it fixed the problem
On 29/05/07, Khaled Chehab [EMAIL PROTECTED] wrote:
I am using asterisk 1.4.4 now and facing a problem with meetme,the code I
was using with asterisk 1.2 is not functioning with 1.4 ,my code is
conf = 222| at meetme.conf
at meet_me_additional
like this
exten = 21,1,MeetMe(21,dq)
exten = 21,2,Playback(beep)
or this
exten = 222,1,GotoIfTime(*|mon-sun|08-08|may-may?223,1)
exten = 222,n,Playback(vm-goodbye)
exten = 222,n,Hangup
exten =
STARTMEETME,1,MeetMe(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN})
exten = STARTMEETME,n,Hangup
exten = h,1,Hangup
exten = 223,1,Set(MEETME_ROOMNUM=222)
exten = 223,n,GotoIf($[${DIALSTATUS} = ANSWER]?READPIN)
exten = 223,n,Answer
exten = 223,n,Wait(1)
exten = 223,n(READPIN),Read(PIN,enter-conf-pin-number,,)
exten = 223,n,GotoIf($[foo${PIN} = foo]?USER)
exten = 223,n,GotoIf($[${PIN} = ]?ADMIN)
exten = 223,n,Playback(conf-invalidpin)
exten = 223,n,Goto(READPIN)
exten = 223,n(ADMIN),Set(MEETME_OPTS=aAwciMs)
exten = 223,n,Goto(STARTMEETME,1)
exten = 223,n(USER),Set(MEETME_OPTS=ciMs)
exten = 223,n,Goto(STARTMEETME,1)
please guide me
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Re: [asterisk-users] auto/forced call
No python code needed . Check .call files at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out On 23/05/07, Brad Sumrall [EMAIL PROTECTED] wrote: Can anyone guide me to a how to on automating a call? I know a little piece of code (normally python) has to be place some where and then a file has to be mv into the spooler. Where do I get the run down? I have a button on another application that sends an email and I want it to also send a text message through asterisk! Brad ___ --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] Call recording filename
I have figured out a way to include dialed number in recorded
voicefile in freepbx . You have to edit
/var/lib/asterisk/agi-bin/recordingcheck
add this lines after $agi=new AGI()
$temp= $agi-get_variable(DIAL_NUMBER);
$agi-verbose(Number to be dialled is -{$temp[data]});
After this you can use variable {$temp[data]} in outfile names ( set
few line below in same file ) . This is only required for freepbx .
On 30/11/06, Vicky [EMAIL PROTECTED] wrote:
No response at all :( . I did a temporary solution . I made cdr mysql to
store unique id into database from this wiki . So i now atleast have
uniquefield common in callfilename and sql records to tally .
Storing the Unique ID
Q: It would appear that the uniqueid field is not being populated in the
MySQL CDR DB. Is this an obsolete field or is a bug?
A: You need to define MYSQL_LOGUNIQUEID at compile time for it to use that
field.
You have two options in /usr/src/asterisk-addons:
1. Add CFLAGS+=-DMYSQL_LOGUNIQUEID to the Makefile.
2. Add a #define MYSQL_LOGUNIQUEID to the top of cdr_addon_mysql.c.
Finally perform the usual make clean, make, make install. Be sure to check
the Makefile for the presence of this flag after having done a CVS update!
You will most probably also want to index the uniqueid field in your cdr
table to improve performance.
On 30/11/06, Nick Hoffman [EMAIL PROTECTED] wrote:
On Wed November 29 2006 05:17, Vicky [EMAIL PROTECTED] wrote:
I am using asterisk along with freepbx . When recording is enabled for a
extension the call record file made in /var/spool/asterisk/monitor
contains information like OUT(extension
number)-(timestamp)-(uniqueid).wav . This can be a big
mess if there are
more than 1000-2000 files in that folder and very hard to locate a call
recording based on call time and extension number who dialled. I need to
put something like outgoing number dialled within call file name instead
of uniqueid .. After watching in console i opened up
/var/lib/asterisk/agi-bin/recordingcheck and saw that
it is setting
callfilename variable with extension number,time,unique id , etc. so i
edited and instead of $uniqueid i put $DIALEDPEERNUMBER ( saw in
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
) but
its just not giving dialed number and hence callfilename doesnt contain
outgoing number . Any suggestions how can i get outgoing call number in
recording file ?
Hi Vicky. Did you receive any responses to your email? I'd be interested
in
anything people suggested.
Cheers,
-- Nick
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[asterisk-users] 0 duration but non-zero billsec in mysql cdr
I was just going through my call records ( stored in mysql database by cdr_MYSQL module ) and saw a record having duration = 0 and billsec of more than 50 seconds . I did a query on cdr where duration billsec and saw that there were infact some 250 records with duration less than billsecond ( table had around 4,00,000 records) . Did anyone came across this ? I also checked csv files and they had same record with duration 0 and higher bill seconds . Happen with both asterisk 1.2.17 as well as 1.2.18 All sip to iax/sip calls . Destination numbers were valid. Dialplan maintained with freepbx . ___ --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] 0 duration but non-zero billsec in mysql cdr
Someone in -biz list pointed out that this could be a freepbx problem so i think i will go check there . @ Salvatore Giudice: how can i intentionally do it ? Damn i need a app that can make sure customer phone doesnt hangup for the time i specify .. even if customer breaks his phone . lol ___ --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] chan_sip seems to be hanging
try soft hangup sip channel name On 02/05/07, Ken Williams [EMAIL PROTECTED] wrote: I posted about this problem last week and thought it was a combination of SIP/ZAP causing issues in Asterisk. Since then I've realized it's only the SIP channel that's hanging. When this happens a call can still come in and hit the IVR, but no one can connect to the server from a SIP client. I tried reloading chan_sip.so today when this occurred, and I tried unloading chan_sip.so but was told the channel was in use. How can I clear SIP connections? With ZAP channels I can use ZAP DESTROY CHANNEL, but I don't see the equivalent for SIP. Any suggestions for tracking down what's causing SIP to hang? My only option as it stands is to shutdown asterisk restart it, I included a piece of the log last week and am willing to do so again if needed. If I can see which SIP channels the server thinks are open when the channel hangs I'm hoping this will allow me to find if it's a common phone or perhaps some dialplan logic gone bad. Thanks, Ken ___ --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] Calls in ulaw, not gsm as desired
Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls from g729 to other codecs or want to record calls ) . Also check your phones config if its support multiple codecs . . On 02/05/07, Rob Schall [EMAIL PROTECTED] wrote: So I reloaded things and had just gsm set for 2 of my polycom 501 phones. However, the logs say No codec found, which leads me to believe that polycom 501 phones can't use gsm. Does anyone have something like this working? If not gsm, is there a better option with these phones over a low bandwidth situation? Rob Ed Nuñez wrote: Reload will reload your sip.conf file! As well as iax.conf, extensions.conf, queues.conf, voicemail.conf, users.conf *From:* [EMAIL PROTECTED] [ mailto:[EMAIL PROTECTED][EMAIL PROTECTED]] *On Behalf Of *Rob Schall *Sent:* Tuesday, May 01, 2007 2:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Calls in ulaw, not gsm as desired I was in the asterisk console and I typed reload. Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: However, even once I reloaded the extensions, its still only using ulaw. You didn't reload the sip config? Maybe that's your problem? -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Make an iso image or a kickstart-Really its too urgent
Why not use a asterisk specific live cd distribution like www.astlinux.org ? It is also installable on usb . You can copy your whole dialplan and settings ( all files in /etc/asterisk ) on a pendrive . On 25/04/07, Khaled Chehab [EMAIL PROTECTED] wrote: Dears its too urgent Can anyone guide me …… I want to put my asterisk system on an iso image like trixbox ,or how to make a. how can I do that ,I am using centos 4.4 final Regards -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
