[Asterisk-Users] cid_num with Asterisk CVS 1.0.12
Hello, How can I access callers number with Asterisk CVS 1.0.12? In new version there are structure cid with field cid_num. And in 1.0.12 only callerid field which is equal to cid_name. I also tried to get it from chan-cdr-src but this is also the same as cid_name or callerid. Mindaugas Kezys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SpanDSP - Still can't send
Hello, I had a major headache trying to send faxes for 2 weeks now. Two things helped me: 1. Correct TIF files (received file not always worked for me) 2. Time synchronization e.g. 'ntpdate time.windows.com'. Now %90 of faxes come nice and clean. (10% - incomplete). If you want test TIF file which works - email me directly. Some people stated that 'echotraining=no' in zapata.conf helps. I haven't noticed. Also |caller had no impact on my fax tests. Strange. I have another problem - I can't send multipage TIF files - only first page arrives. Nothing helps - no info in Google. Sincerely, Mindaugas Kezys -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Thursday, February 24, 2005 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SpanDSP - Still can't send Fair point. The FAX apps used to display a note about this when you checked their parameters. That seems to have disappeared from the source code of both rxfax and txfax. I just put it back in. Regards, Steve Rod Bacon wrote: From my personal experience, the 'weird ideas' come from a lack of consistent documentation. The caller option, and the use thereof is not clearly explained anywhere that I can find, and examples of spandsp that are floating around the WiKi (among other places) erroneously leave this option out. In any case, I still can't send faxes... with or without the option. - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 24, 2005 11:02 AM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send Where do these weird ideas come from? :-) Why on earth would it relate to the dial plan? What difference can you imagine the dial plan might make? If you want txfax to behave as the calling, rather than the answering, side you need to specify the caller option. Don't confuse caller with sender. The machine which makes a call can be either the one sending a FAX, or one picking up a FAX by polling. That is the reason the caller option exists. Regards, Steve Rod Bacon wrote: My understanding is that this is only required when using it inside a dialplan. Eg, the extension answers, then switches to fax originator mode. For testing, I am using txfax within a sample.call file which I drop into /var/spool/asterisk/outgoing, as per latest docs. Listening at the receiving fax, the tones sound right, although I must confess that I don't actually speak fax. Does anyone have any ideas about debugging SpanDSP? - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 23, 2005 6:28 PM Subject: Re: [Asterisk-Users] SpanDSP - Still can't send On Wed, 23 Feb 2005, Rod Bacon wrote: No matter which version of SpanDSP I use, with which version of libtiff, Asterisk, ... I simply cannot send faxes. Did you remember to add the caller option to txfax? Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk realtime
MOR Billing comes in PRO and FREE versions. FREE version is OpenSource. It supports Realtime and GUI is written in RoR. It has own app_mor.so written in C and is very fast. You can check it: http://www.kolmisoft.com/mor Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Thursday, August 16, 2007 10:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] GUI for Asterisk realtime Thanks. I had googled as well and found basically the same links. We are building a DUNDi/Realtime cluster and need gui management. Mike.. http://www.bicomsystems.com/products/ Senad ___ --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] Multiple servers using realtime
That's a good note about MySQL replication. You can use it to remove point-of-failure which currently is your DB server. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Collier Sent: Thursday, August 23, 2007 12:21 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Multiple servers using realtime I use a centralized database (with replication) for several servers, and it works very well. I keep all the mysql traffic on a separate network from the SIP traffic. It makes it easy to add capacity. If you are doing all the mySQL on one box anyway, I don?t see any adavantage to using separate databases. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Peder @ NetworkOblivion Enviado el: miercoles, 22 de agosto de 2007 19:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Multiple servers using realtime I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --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] Multiple servers using realtime
Users register to (Open)SER which uses same DB as all Asterisk nodes. Asterisk Realtime engine lets change data in only one database to make changes global. (Open)SER does load-balancing and fail-over. You can even put second (Open)SER server in case first dies and use DNS SRV to make it active. Database (DB) can be on same machine, but it better should be dedicated to only DB to serve only queries from all nodes. Possible to use MySQL Replication and have same DB on all nodes, which will save some processing power. But it's harder to manage. There're tools, choice is yours how you use them. Regards/Pagarbiai, Mindaugas Kezys VoIP Billing Solutions http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Wednesday, August 29, 2007 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple servers using realtime On Tue, 2007-08-28 at 19:59 -0600, Edgar Guadamuz wrote: I have a confusion about using SER for balancing the load across the Asterisk boxes. The doubt is: once a user registers in a Asterisk box, all the calls from or to him are going to be done by the same Asterisk server or can a user make a call by one Asterisk server and then make another call by other Asterisk server? I think the user registers with the SER box. With loadbalancing an outgoing call can go through different Asterisk boxes: call #1 -- SER box #1 -- Asterisk box #1 -- destination call #2 -- SER box #1 -- Asterisk box #2 -- destination Regards, Patrick On 8/28/07, Dovid B [EMAIL PROTECTED] wrote: We have a similar set up. I would recommend also using SER and load balancing so you can load balance your calls out between your asterisk box's. ___ --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] Prepaid Billing: A2Billing, AstBill, ASTCC
You can try MOR FREE billing system for Asterisk. LiveCD can be downloaded from: http://www.kolmisoft.com/mor/index.php?option=com_contenttask=viewid=73 Regards/Pagarbiai, VoIP Billing Solutions Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Monday, August 27, 2007 1:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prepaid Billing: A2Billing, AstBill, ASTCC Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460 Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --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] Voicemail and fax detect
You can try this: http://www.voip-info.org/wiki-NVFaxDetect Regards/Pagarbiai, Mindaugas Kezys VoIP Billing Solutions http://www.kolmisoft.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Wednesday, August 29, 2007 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voicemail and fax detect Hi, Has anyone experienced bundling fax detection and voicemail ? My goal is to listen to incoming calls beginning before forwarding them to spandsp or voicemail, accordingly. This feature is not business critical as casual collective fax numbers remain available for important faxes. This is more for private or sensitive fax someone can receive from time to time. What did you experienced (user satisfaction, ...) ? Best regards ___ --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] special kind of billing
You can try MOR: www.kolmisoft.com/mor It does what you need. It does it even in FREE version. PRO version costs _many_ times less then other not free solutions mentioned in this thread. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kate Kretz Sent: Wednesday, September 05, 2007 7:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] special kind of billing Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Cheers, Kate ___ 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] Silencing VoiceMail() app in * 1.4.10
Hello, Just find this file in /var/lib/asterisk/sounds and change it to anything you like. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Horwich IT Services (Godwin Stewart) Sent: Friday, March 07, 2008 10:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Silencing VoiceMail() app in * 1.4.10 Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten = 2,1,Playback(/media/asterisk/answerphone-en) exten = 2,n,VoiceMail(2000,s) exten = 2,n,Playback(/media/asterisk/thankyou-en) exten = 2,n,Hangup() The 's' option to VoiceMail() silences the prompt, leaves the beep just before going into 'record' mode, but also plays back auth-thankyou after the user hits the # key. How can I suppress playback of auth-thankyou at the end or get VoiceMail() to play back a different file? Thanks in advance, -- Godwin Stewart - Horwich IT services ___ -- 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] Silencing VoiceMail() app in * 1.4.10
Hello, Then you can change channel language in front of VoiceMail() app and in appropriate place put auth-thankyou file which is recorded/made by you. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Horwich IT Services (Godwin Stewart) Sent: Friday, March 07, 2008 1:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Silencing VoiceMail() app in * 1.4.10 On Fri, 7 Mar 2008 12:10:37 +0200, Mindaugas Kezys [EMAIL PROTECTED] wrote: Just find this file in /var/lib/asterisk/sounds and change it to anything you like. But that will break other applications that use the auth-thankyou sound, Authenticate() for a start (which I use elsewhere in order to remote check the voicemailbox). -- Godwin Stewart - Horwich IT services ___ -- 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] T.38 SIP Issues
Hello, This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Thursday, March 13, 2008 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 SIP Issues Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. ___ -- 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] T.38 SIP Issues
Hello, Higher speeds then 9600kbps are not permited by patents. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Friday, March 14, 2008 3:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T.38 SIP Issues Has someone submitted a bugreport regarding enabling 9600kbps fax? I always wonder why it would never negociate 14400kbps... when it did work a single page on fine resolution would take 4 minutes. Thank you very much for that link. I knew there had to be more possible configurations for T.38. I will give it a try... but I think I can get away without patching chan_sip.c, no? that just seems to enable higher bitrates. And Linksys SPA2102 is one of the exact devices I have in my lab. On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys [EMAIL PROTECTED] wrote: Hello, This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem Helge Sent: Thursday, March 13, 2008 5:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] T.38 SIP Issues Is there any trick to getting T.38 fax to work with SIP? I had it working and one day with no changes *poof* it stopped working and hasn't worked for months. The only common factor is Asterisk 1.4.x (always try to use the latest version) and NAT. I've tried: -Linksys ATA -Grandstream ATA -Audicodes ATA All do the same thing. Call connects, hear the first 2sec of fax tone and then just silence, but the call usually stays open. I've tried two T.38-capable providers. I've tried two different routers: -Linksys WRT54GS running DD-WRT (Linux) -Dell Optiplex 170L running PFSense (BSD) Different Linux distros on the servers: -SuSE 64bit -RHEL 32bit -SuSE 32bit Is there any magic to get this to work? As far as I can tell the only possible config option is t38pt_udptl = yes which I have set under [general] the peer. ___ -- 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] Redundant Voicemail
Hello, This can help: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ein Bielaczyc Sent: Monday, March 17, 2008 3:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Redundant Voicemail Forgive me if this has been covered before. I did search but I was unable to find a reference. I am curious to know more about the possibility of using SQL to store voicemail as well as having more than one voicemail system accessing a central SQL database. Any information would be appreciated. Thank you all, in advance. -- Ein Bielaczyc [EMAIL PROTECTED] NOTICE: This E-mail (including attachments) is covered by the Electronic Communications Privacy Act, 18 U.S.C.2510-2521, is confidential and may be legally privileged. If you are not the intended recipient, you are hereby notified that any retention, dissemination, distribution or copying of this communication is strictly prohibited. Please reply to the sender that you have received the message in error, then delete it. ___ -- 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] How is uniqueid computed
Hello, Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 ) If call is transfered or it is leg2 then: Uniqueid = (call initiation time in unix time format) . (call count since asterisk restart / 2 + 1) This is from observations, i can be mistaken. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing Solution for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, March 18, 2008 6:12 PM To: asterisk-users Subject: [asterisk-users] How is uniqueid computed Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- 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] How to obtain SIPCHANINFO variables within custom application?
Hello,
How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough
variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my custom
application, because there's no chan_sip.h and I can't include it into my
application (maybe there's other way?).
I can do like this:
exten = _X,1,Set(PEERIP=${SIPCHANINFO(peerip)})
exten = _X,2,custom_app
and read PEERIP with
pbx_builtin_getvar_helper, but that's not an option for me.
Any help?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
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Re: [asterisk-users] How to obtain SIPCHANINFO variables within custom application?
Thank you!
You saved my day!
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: Tuesday, March 25, 2008 5:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to obtain SIPCHANINFO variables
within custom application?
On Tuesday 25 March 2008 07:51:13 Mindaugas Kezys wrote:
How can I get peerip, recvip, from, uri, useragent, peername,
t38passthrough variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my
custom
application, because there's no chan_sip.h and I can't include it
into my
application (maybe there's other way?).
I can do like this:
exten = _X,1,Set(PEERIP=${SIPCHANINFO(peerip)})
exten = _X,2,custom_app
char buf[80];
pbx_substitute_variables_helper(chan, ${SIPCHANINFO(peerip)}, buf,
sizeof(buf));
BTW, this is exactly how res_config_curl works.
--
Tilghman
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Re: [asterisk-users] rxfax crashes Asterisk (segmentation fault)
Hello, Rxfax from agx-ags-addons always crashes for us also. You can download apps we use from: http://193.138.191.205/packets/fax_apps_asterisk14.tgz Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced VoIP Billing From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mark morreny Sent: Friday, April 04, 2008 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rxfax crashes Asterisk (segmentation fault) Hi, I am using spandsp-0.0.4, tiff-3.8.2, and agx-ags-addon with Asterisk 1.4.18. Everytime rxfax executes, Asterisk crashes: -- Executing [EMAIL PROTECTED]:1] Set(Zap/2-1, FAXFILE=/var/spool/asterisk-fax/1207322398.0.tif) in new stack -- Executing [EMAIL PROTECTED]:2] RxFAX(Zap/2-1, /var/spool/asterisk-fax/1207322398.0.tif) in new st ack [Apr 4 23:20:35] NOTICE[23925]: chan_iax2.c:6025 update_registry: Restricting registration for peer ' iaxmodem' to 60 seconds (requested 50) [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Pages transferred: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size: - 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image resolution- 1209075756 x -1221451281 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Transfer Rate: - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Bad rows- 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Longest bad row run - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Compression typea st_speech_unregister [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: Image size (bytes) - 1209075756 [Apr 4 23:21:03] DEBUG[23953]: /usr/src/agx-ast-addons/app_rxfax.c:81 xast_log: = = Segmentation fault Is rxfax supposed to be working? What could have caused this problem? Thanks, Mark ___ -- 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 1.4.19 crash with Realtime using SIP peers
Hello,
Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
gdb asterisk /tmp/coreXXX shows:
Program terminated with signal 11, Segmentation fault.
#0 0xb6148968 in find_peer (peer=0xb6042768 test, sin=0x0, realtime=1) at
chan_sip.c:2547
2547if (!(hp =
ast_gethostbyname(tmp-value, ahp)) || (memcmp(hp-h_addr, sin-sin_addr,
sizeof(hp-h_addr {
Sorry, I have no time to read manual how to correctly put this into bug
tracker.
Back to 1.4.18.1
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
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Re: [asterisk-users] *21*number # diverting
Google is your friend: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced VoIP Billing Solution From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gilbert saunders Sent: Wednesday, April 09, 2008 4:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] *21*number # diverting hi me again im new at asterisk and really need some good tutoring on asterisk and call forwarding i dont understand it at all pls help i have attached my extensions.conf file if someone would be so kind to look at it and tell me what code i must enter to make *21*number diverting and #21# undiverting possible __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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
Re: [asterisk-users] problem with Asterisk 1.4.19 - accountcode dissapearing
As far as I noticed - this issue is not 1.4.19 only. Same thing happens on
all Asterisk versions.
Set your own variable before transfer:
Exten = , Set(__MYACC=${CDR(accountcode)})
And use ${MYACC} in other (transfered) calls.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, April 15, 2008 3:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] problem with Asterisk 1.4.19 - accountcode
dissapearing
Hi,
I have a big issue during transfers (using Polycom phones, but I don't think
that's relevent) with Asterisk 1.14.19. Basically, the value contained in
${CDR(accountcode)} dissapears.
Here is the relevant code snippet:
--
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS CORRECT AND IS
EQUALS TO THE ACCOUNTCODE SPECIFIED MUCH EARLIER IN THE DIALPLAN
exten = _X!.,n,Gotoif($[${i} = 1]?$[${PRIORITY}+2])
;DIAL ALL MAC PHONE ASSOCIATED WITH THIS EXTENSION SIMULATENOUSLY
exten =
_X!.,n,Dial(${mac_dial_string:0:$[${LEN(${mac_dial_string})}-20]}|${sip_phon
es_ring_time}) ;remove least 7 characters, thos
e are left there by the invalid last SQL fetch
exten = _X!.,n,Set(i=0)
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS EMPTY, and so
is this variable if I use it in any way.
When I dial an extension and it hits this diaplan, it works fine. But if I
dial an extension, answer and then transfer (using Polycom phones) to an
extension using this dialplan I lose the accountcode where specified in the
code. It's empty. How can ${CDR(accountcode)} lose it's value for no
reason in those two seemingly innocent diaplan lines?
Below is the CLI output if it's useful:
-- Executing [EMAIL PROTECTED]:22]
NoOp(SIP/0004f2134384-1-097fb4e8, 1234567890) in new stack ;THIS IS THE
ACCOUNTCODE
-- Executing [EMAIL PROTECTED]:23]
GotoIf(SIP/0004f2134384-1-097fb4e8, 0?25) in new stack
-- Executing [EMAIL PROTECTED]:24]
Dial(SIP/0004f2134384-1-097fb4e8, SIP/0004f2134384-3|8) in new stack
-- Called 0004f2134384-3
-- SIP/0004f2134384-3-099947b0 is ringing
== Spawn extension (generic-extensions-db, 705, 24) exited non-zero on
'SIP/0004f2134384-1-097fb4e8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back from
192.168.1.6
-- Nobody picked up in 8000 ms
-- Executing [EMAIL PROTECTED]:25]
Set(SIP/0004f212ae63-1-099700a8, i=0) in new stack
-- Executing [EMAIL PROTECTED]:26]
NoOp(SIP/0004f212ae63-1-099700a8, ) in new stack ;MISSING ACCOUNTCODE
IS HERE
Mick
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Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
AGX-Addons crashes Asterisk for us. Working solution (on 100+ servers we installed): - apt-get -y install g++ libtiff4 libtiff4-dev patch autoconf automake libtiff-tools cd /usr/src wget http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20080402.tar. gz tar xzvf spandsp-20080402.tar.gz cd /usr/src/spandsp-0.0.4 ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig cd /usr/src wget http://193.138.191.205/packets/fax_apps_asterisk14.tgz tar xzvf fax_apps_asterisk14.tgz cd /usr/src/fax_apps make make install Restart Asterisk. Voila! Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Sent: Wednesday, April 16, 2008 5:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing No progress at all. Version from Debian/Lenny repository still crashes and I'm not able to compile AGX. It gives out a long list of error messages. Some unsatisfied dependencies...? I Can't experiment for a while after unwanted night-time visit of fire-fighters :-( I have to let everything dry and clean out of sand and drywall pieces :-( Martin - Original Message - From: Justin Newman mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: 11. dubna 2008 13:00 Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing Did this just start happening with the 1.4 tree? Have you made any progress on getting it resolved? Justin Newman Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: Let's be more specific here, folks: What version numbers? Asterisk, spandsp, agx-addons / rx-tx-fax? Asterisk: yesterday's 1.4 SVN SpanDSP: tried with pre 15, 16 and 18 AGX-Addons: tried with 1.4.5 and svn trunk rx/txfax: supplied by AGX Addons - although they seem to build the files and stick them into the modules directory, rather than adding to the apps directory and modifying the Makefile. i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5 linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax when i enable faxdetect in zapata.conf. since then it disabled faxdetect and use nvfaxdetect function in dialplan, it works fine afterward. also it seems to works fine using regular 32bit kernel. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 search=0xD6506D20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] keep one line open
Check who is dialing this line by CallerID, if it is not your user - just drop the call. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of gilbert saunders Sent: Thursday, April 17, 2008 8:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] keep one line open hi i have multiple lines going to my asterisk box etc 0282549087 , 028 3659874 , 0285469658 etc. is it possible to keep users from using the 0282549087 line always open that it only allows a certain user to make outgoing calls on it? _ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try http://us.rd.yahoo.com/evt=51733/*http:/mobile.yahoo.com/;_ylt=Ahu06i62sR8H DtDypao8Wcj9tAcJ%20 it now. ___ -- 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] RTCP stats
Hello, Is here an easy way to get RTCP Stats in channel variables after the call ends? Or source should be edited to accomplish this? I would like to know this before developing this feature. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX ___ -- 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 in Production ?
Hello, Our company did 200+ installations around the globe and had no issues with stability with correct Asterisk version. We used most of 1.4. As far as I remember 1.4.16 has some nasty bugs along with 1.4.19.x (SIP + realtime). So current stable is 1.4.18.1 (for us). For load check: http://wiki.kolmisoft.com/index.php/How_fast_MOR_can_perform It shows how our billing application performs on top of Asterisk (2049 channels) and we can push it even further with some improvements. We DO NOT RESTART our Asterisk installations daily or weekly. They work for months. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Benoit Plessis Sent: Tuesday, May 06, 2008 2:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk in Production ? Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards ___ -- 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 and OOH323
Not sure about OOH323, but H323 can: http://www.voip-info.org/wiki/view/Asterisk+RealTime+H323 Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: Saturday, June 21, 2008 10:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Realtime and OOH323 Can Realtime be used with OOH323 ala sip_buddies? ___ -- 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] Major problem with 1.4.21 asterisk
Same here. Some of our clients upgraded from 1.4.18.1 to 1.4.21. After some time CLI stops responding and no calls are possible. Killall -9 is the only way to solve (get out) of this situation till next time it hangs. Example CLI screenshot: http://193.138.191.205/packets/asterisk1.4.21_noresponse.jpg Back to 1.4.18.1 (1.4.19.x is even more broken: http://lists.digium.com/pipermail/asterisk-users/2008-April/209342.html). Regards, Mindaugas Kezys http://www.kolmisoft.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Wednesday, June 25, 2008 7:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Major problem with 1.4.21 asterisk Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks 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 -- 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
[asterisk-users] How to pass variable between 2 Asterisk servers over IAX2
Hello, Anybody can advice how to pass variable between 2 Asterisk servers over IAX2? With SIP I can use SipAddHeader. How do to the same with IAX2? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com ___ -- 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] How to pass variable between 2 Asterisk servers over IAX2
Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Friday, June 27, 2008 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to pass variable between 2 Asterisk servers over IAX2 On Friday 27 June 2008 10:07:18 Mindaugas Kezys wrote: Anybody can advice how to pass variable between 2 Asterisk servers over IAX2? With SIP I can use SipAddHeader. How do to the same with IAX2? In 1.6, with IAXVAR(). -- Tilghman ___ -- 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
[asterisk-users] MOR billing and routing 0.6 released
Hello, We are proudly to present new version of our billing and routing system MOR v0.6 More info: http://www.voip-info.org/wiki/view/MOR Regards, Mindaugas Kezys http://www.kolmisoft.com ___ -- 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] Problems w/Asterisk Realtime + MySQL + SIP
Hi, Try to delete whole column 'md5secret' from DB peers table. Leave only 'secret'. And try then. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Walter Stanish Sent: Monday, July 21, 2008 8:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems w/Asterisk Realtime + MySQL + SIP [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. It looks like Asterisk is unhappy with the SIP REGISTER request coming from your softphone for some reason. It's very strange that it's occurring for two different softphones though. Trun on SIP debugging by typing sip debug on your Asterisk console and then post up the 4 SIP messages invloved in the register transaction so we can take a look and spot why it could be getting rejected. Sure. Here's what happens when kphone starts up: == --- SIP read from 192.168.0.25:5060 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C CSeq: 35 REGISTER To: Walter sip:[EMAIL PROTECTED] Expires: 900 From: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: Walter sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER black*CLI - --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.25 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 35 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK5760BF8C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED];tag=as59de1023 Call-ID: [EMAIL PROTECTED] CSeq: 35 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7864265a Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) == Kphone prompts for a password, then the following occurs. == --- SIP read from 192.168.0.25:5060 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C CSeq: 36 REGISTER To: Walter sip:[EMAIL PROTECTED] Authorization: Digest username=walter, realm=asterisk, nonce=7864265a, uri=sip:192.168.0.2, cnonce=abcdefghi, nc=0001, response=10a7024959390c04b4d09c708fac6130, opaque=, algorithm=MD5 Expires: 900 From: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.2 Event: registration Allow-Events: presence Contact: Walter sip:[EMAIL PROTECTED];transport=udp;methods=INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER - --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.25 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 36 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.25:5060 --- SIP/2.0 403 Forbidden (Bad auth) Via: SIP/2.0/UDP 192.168.0.25;branch=z9hG4bK36B0646C;received=192.168.0.25 From: Walter sip:[EMAIL PROTECTED] To: Walter sip:[EMAIL PROTECTED];tag=as59de1023 Call-ID: [EMAIL PROTECTED] CSeq: 36 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 [Jul 22 00:59:38] NOTICE[2414]: chan_sip.c:15049 handle_request_register: Registration from 'Walter sip:[EMAIL PROTECTED]' failed for '192.168.0.25' - Wrong password Scheduling destruction of SIP dialog '[EMAIL
[asterisk-users] Whitepaper: How and to whom sell VoIP
Hello, Based on our own and our clients' experience we compiled short manual: How and to whom sell VoIP Hope it can be useful to some of you also. You can download it from our site: http://www.kolmisoft.com Regards, Mindaugas Kezys http://www.kolmisoft.com ___ -- 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] New GUI for Realtime Asterisk - RAGUI
Congratulations with the release! I'm curious also about the statement on your page: Realtime Asterisk uses the MySQL relational database to access dialplan, extension and configuration data. This allows for dynamic additions and changes to users, extensions and dialplans without having to restart or reload the system. What version of Asterisk are you using? From my experience starting from 1.4.19 Asterisk Realtime is completely broken: 1. http://bugs.digium.com/view.php?id=12362 2. http://bugs.digium.com/view.php?id=12925 3. http://bugs.digium.com/view.php?id=12921 Also how do you go about changing details for device in DB and not using sip realtime prune PEER + 'sip reload'? Without that your changes to devices are not active. Good luck! Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Clark Sent: Thursday, August 14, 2008 12:07 AM To: asterisk-users@lists.digium.com; Commercial and Business-Oriented Asterisk Discussion Subject: [asterisk-users] New GUI for Realtime Asterisk - RAGUI Our company, WebPoint IT Solutions has just released an open source (GPL V2 license), Ruby on Rails based gui manager for Realtime Asterisk called RAGUI. RAGUI is definitely a work in progress and has rough edges, but we expect to polish it up in the upcoming weeks and months. All comments, contributions, and criticisms are welcomed! Here are the links: Sourceforge: http://sourceforge.net/projects/ragui/ Website: http://www.ragui.net Enjoy! Mike Clark WebPoint IT Solutions ___ -- 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] strange h323 delay issue
Hi, Try downgrade to Asterisk 1.4.18.1. It works for us perfectly with H323. Following versions has nasty bugs, not actually related to H323, but who knows, maybe it will help to downgrade. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys Sent: 2008 m. spalio 18 d. 22:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] strange h323 delay issue Hello, I have a strange h323 issue. After executing command Dial(SIP/333-0d1dfe00, H323/[EMAIL PROTECTED]|5|tT) at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what asterisk (h323 channel) was doing for 40 sec??? What reasons could invoke this problem? I haven't any problems with SIP channels. My versions: asterisk-1.4.21.1 asterisk-addons-1.4.6 openh323_v1_18_0 pwlib_v1_10_0 My h323.conf configurations: [general] port = 1720 bindaddr = 192.168.1.165 tos=lowdelay disallow=all allow=g729 dtmfmode=rfc2833 gatekeeper = DISABLE AllowGKRouted = no AcceptAnonymous = no context=from-trunk [ccg] type=friend context=from-trunk host=192.168.1.163 port=1720 disallow=all ;allow=alaw ;allow=ulaw allow=g729 fastStart=yes h245Tunneling=yes A full log: [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/333-0d1d8fb0, H323/[EMAIL PROTECTED]|5|tT) in new stack [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: type=H323, format=8, [EMAIL PROTECTED] [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Extension: 361737052390920 Host: ccg [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Calling to [EMAIL PROTECTED] on H323/ccg-2 [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Requested transfer capability: 0x00 - SPEECH [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Placing outgoing call to [EMAIL PROTECTED]:1720, 101 [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Making call to [EMAIL PROTECTED]:1720 without gatekeeper. [Oct 18 22:32:23] VERBOSE[18236] logger.c: Using 192.168.1.165 for outbound call [Oct 18 22:33:03] VERBOSE[18236] logger.c: == New H.323 Connection created. [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- root is calling host [EMAIL PROTECTED]:1720 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Call token is ip$localhost/6453 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Call reference is 6453 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- DTMF Payload is [pt=101] [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Called [EMAIL PROTECTED] [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Setting capabilities for connection ip$localhost/6453 [Oct 18 22:33:03] VERBOSE[18238] logger.c: Setting capabilities to 0x100 (g729) [Oct 18 22:33:03] VERBOSE[18238] logger.c: Capabilities in preference order is (g729) [Oct 18 22:33:03] VERBOSE[18238] logger.c: Allowed Codecs: [Oct 18 22:33:03] VERBOSE[18238] logger.c: Table: [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729A 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729 2 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/hookflash 3 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/RFC2833 4 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/dtmf 5 [Oct 18 22:33:03] VERBOSE[18238] logger.c: Set: [Oct 18 22:33:03] VERBOSE[18238] logger.c:0: [Oct 18 22:33:03] VERBOSE[18238] logger.c: 0: [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729A 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729 2 [Oct 18 22:33:03] VERBOSE[18238] logger.c: 1: [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/hookflash 3 [Oct 18 22:33:03] VERBOSE[18238] logger.c: 2: [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/RFC2833 4 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/dtmf 5 [Oct 18 22:33:03] VERBOSE[18238] logger.c: [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Capabilities for connection ip$localhost/6453 is set [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Created RTP channel [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Setting NAT on RTP to 0 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Sending SETUP message [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Transmitting RFC2833 on payload 101 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Started logical channel: sending G.729A [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- channelsOpen = 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c: External RTP
RE: [asterisk-users] Some problems with mysql CDR
Hello,
Is your userfield type varchar(255)?
Also check if you edited the cdr_addon_mysql.c and Make file to tell
cdr_addon_mysql.c to store uniqueid as outlined here:
http://www.voip-info.org/wiki-Asterisk+cdr+mysql
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate
entry '' for key 1
This happens when you have UNIQUE for some field and insert same record twice.
In order to help please paste your [cdr] table structure.
I'm sure it's not a bug but misconfiguration which can be solved easily.
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP ServicesSolutions
MOR - FREE Open Source billing for Asterisk
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Martin
Sent: Monday, May 14, 2007 10:34 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Some problems with mysql CDR
Hello,
We have finally upgraded to Asterisk 1.4, however we've run into two issues
that weren't occurring before the upgrade.
Issue #1: We're an outgoing call center and need to record all calls. We use
the uniqueid field in the CDR to match with the recording, which we labeled
with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct
in the CDR. Here is the manager event for a call:
Event: Cdr
Privilege: call,all
AccountCode: 6384106:MMI-Y:200705081051010077
Source: 00
Destination: 6398714109927773
DestinationContext: outbound
CallerID: 00
Channel: Zap/15-1
DestinationChannel: SIP/teliax-081ed5b0
LastApplication: NoOp
LastData:
StartTime: 2007-05-08 10:51:04
AnswerTime: 2007-05-08 10:51:05
EndTime: 2007-05-08 11:01:56
Duration: 652
BillableSeconds: 651
Disposition: ANSWERED
AMAFlags: DOCUMENTATION
UniqueID: 1178635864.1510
UserField:
And for that record in the database:
'calldate' '2007-05-08 10:51:04'
'clid' '00'
'src' '00'
'dst' '6398714109927773'
'dcontext' 'outbound'
'channel' 'Zap/15-1'
'dstchannel' 'SIP/teliax-081ed5b0'
'lastapp' 'NoOp'
'lastdata' '',
'duration' 652,
'billsec' 651,
'disposition' 'ANSWERED',
'amaflags' 3,
'accountcode' '6384106:MMI-Y:200705081051010077',
'uniqueid' '51010077',
'userfield' '',
'MMI_field' 'not found'
Issue #2: When a call is not answered, a record of that call is written to the
database, but uniqueid is left blank. The next time a call isn't answered,
Asterisk complains:
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate
entry '' for key 1
I haven't found any other information regarding these errors. I am just
wondering if they are bugs. Any insight would be appreciated!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679
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RE: [asterisk-users] Originate and ForkCDR()
Hello, I guess you have something like this: ACTION: Originate Channel: Local/1234 Exten: 4321 Priority: 1 Context: blabla And in [blabla] Exten = 4321,1,Dial(something. Instead use magic /n setting with Local channel. See here: http://www.voip-info.org/wiki/view/Asterisk+local+channels And do your originate like this: ACTION: Originate Channel: Local/[EMAIL PROTECTED]/n Exten: 4321 Priority: 1 Context: blabla And in [blabla] Exten = 4321,1,Dial(Local/something/n) I guess you got the idea. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com VoIP SolutionsServices MOR - FREE Open Source billing for Asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Federico Cabiddu Sent: Tuesday, May 15, 2007 2:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Originate and ForkCDR() Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the answer field is always empty, billsec field is 0 and disposition field is always set to NO ANSWER. Is there something I'm missing? Thanks, Federico -- Federico Cabiddu RD Software Engineering Abbeynet S.p.A. - www.abbeynet.com http://www.abbeynet.com phone: +39 070 2339349 http://www.federico_cabiddu.sitofono.it ___ --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] voice recording on legacy PBX
Hi, Connect you Asterix box in the middle of call-flow and you will be able to record all calls. PSTN - Asterisk - Legacy PBX - Phones Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com VoIP ServicesSolutions MOR - FREE Open Source billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, May 16, 2007 1:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] voice recording on legacy PBX Hi, Is it possible to use Asterisk to record or monitor all conversation on standard PSTN PBX ? ASLAY ___ --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
[asterisk-users] Advanced Billing System for Asterisk - MOR v0.4 released
Hello, We are proudly to present new version of our billing system MOR v0.4 What's new in MOR FREE v0.4 * Extended stability and reliability * Extended configuration options for clients and providers * User blocking * Prepaid support * Increased security * New tariff/rating engine * Registration * PayPal integration * Authorization by IP * Extended calls view What's new in MOR PRO v0.4 * Extended Calling Card engine * CDR import from CSV file * Device grouping * Country Stats * Providers' Stats * Auto-Dialer * Callback * Click2Call * Invoice generation * Custom Rates * Localization More info in: http://www.kolmisoft.com/mor Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New release of billing and routing software MOR
Hello, We are glad to announce new release of our advanced billing and routing package for Asterisk - MOR v0.7 It is complete solution for VoIP billing and routing for advanced and start-up telecoms, carriers, voip calling card operators and ISPs. Demo available online, as LiveCD or as InstallCD. Contact us for more details. More info: http://www.kolmisoft.com What is new in this version: * Call Routing by priority (Manual LCR) * LCR/Tariff change based on call prefix * PBX Functions - small functions which extends functionality of MOR PRO * PDF UTF8 support * More statistical data * New permission system * Accountant role * CallerID Manipulation: * Localization/Provider Rules * CallerID change on Forward * SIP debug system * New payment gateways: LinkPoint and CyberPlat * Google Maps integration to show Active Calls on the map!!! * IVR system * Limit calls per provider/did/user/device basis * User/Device/DID import from files * Send invoices by email in batches * NO ANSWER/BUSY interpretation for providers * Currency engine rework - automatic update from web Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] H323 crashes Asterisk on high load
Hello, Asterisk 1.4.18.1 PWlib 1.10.0 Openh323 1.18.0 ../asterisk/channels/h323 compiled from source. Under high load H323 crashes and kills Asterisk, debug shows: (gdb) bt #0 0x007a2b18 in strcmp () from /lib/libc.so.6 #1 0x014478a1 in find_call_locked (call_reference=13, token=0xa1cc570 ip$81.192.72.46:7768/13) at chan_h323.c:1148 #2 0x01449f07 in cleanup_connection (call_reference=13, call_token=0xa1cc570 ip$81.192.72.46:7768/13) at chan_h323.c:2290 #3 0x0145a724 in MyH323EndPoint::OnConnectionCleared () from /usr/lib/asterisk/modules/chan_h323.so #4 0x00e604f1 in H323Connection::OnCleared () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #5 0x00e721d1 in H323EndPoint::CleanUpConnections () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #6 0x00e722fe in H323ConnectionsCleaner::Main () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #7 0x005fd6e5 in PThread::PX_ThreadStart () from /usr/local/lib/libpt_linux_x86_r.so.1.10.0 #8 0x0088446b in start_thread () from /lib/libpthread.so.0 #9 0x00804dbe in clone () from /lib/libc.so.6 Server 2x XEON quad core and 4g DDR crashes on 110-120 simm. H323 calls. Anybody experienced same situation? Maybe there is some fix? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Using Asterisk to measure call quality: Introducing Recqual
On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote: Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: Looks very interesting. After reading all available info I have two questions before testing: 1. Who/what answers the calls at the other end? I guess real live traffic should be sent through this Asterisk server? 2. How many calls you had made to to diagnose your problems? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Background stress test
Hello, We did small test with sipp to test Asterisk Background command capability. Our goal was 700 sim. calls on HP Proliant DL160 G5 E5405 1 x Quad Core Xeon 2Ghz 2 Gb RAM Asterisk 1.4.18.1 Centos 5.2 We reached more then 1000 when our network (100mbps) become a bottleneck. As we achieved our goal - no further testing was performed. As conclusion - we are very happy with Asterisk in this case. If somebody is interested - more details are here: http://wiki.kolmisoft.com/index.php/Asterisk_Background_performance_test Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] H323 stress test
Hello, We made small stress-test for H323. Test shows that H323 protocol is heavyweight compared with SIP. More details: http://wiki.kolmisoft.com/index.php/H323_pass-through_test Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Caller Hangup detection
Hello, Is here any dial plan variable which could help me to identify that call was dropped (when still not connected) by caller? HANGUPCAUSE returns 0 DIALSTATUS returns NOANSWER How to identify such situation? Related question - how to know which end (caller or callee) ended the call first after call was answered? Thank you. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Dial() application 'g' option
How to determine which channel hung up first?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: 2009 m. vasario 22 d. 04:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial() application 'g' option
On Saturday, February 21, 2009, Philipp Kempgen wrote:
To be quite precise the documentation says
---cut---
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
---cut---
So I would not expect the g option to have any effect if the
*source* channel hangs up.
I guess you should do any kind of logging or post-hangup calculations
in the h extension.
Thanks. I did wonder about that but carried out some experiments that
suggested it didn't matter which channel hung up first. I have two SIP
geographical numbers with different providers and I tried ringing one
from the other and got the same result no matter which handset I hung
up first.
Unfortunately, by the time the call gets to the h extension, the
original dialled number in ${EXTEN} is changed to h - so I won't be
able to carry out the desired logging there. Also, I suspect that
${DIALEDTIME} and ${ANSWEREDTIME} might be lost. That said, I'm only
interested in recording the accumulated time for outgoing calls via
one SIP trunk, so if I can tie that down with a channel name...
Some further experimentation is in order!
Thanks again,
--
Geoff
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Re: [asterisk-users] VoIP Information in CDRs
I'm trying to find the answer to the same question:
4. Find out who hangedup an answered call.
HANGUPCAUSE = 16
DIALSTATUS = ANSWERER
In both cases, so these variables does not help.
Can anybody help with this issue? Should be pretty simple to detect which
part hanguped the call first.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Costa Tsaousis
Sent: 2009 m. vasario 21 d. 17:30
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VoIP Information in CDRs
Hi,
I am trying to find a way to add the following info in CDRs (with asterisk
1.4.23.1):
1. Codec used
2. RTP QoS statistics
3. RTP IP of remote host
4. For answered calls, the peer that requested to end the conversation
I have managed to get 1 and 2 for the caller, like that:
exten = h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}
Codec=${CHANNEL(audioreadformat)}/${CHANNEL(audiowriteformat)}/${CHANNEL(aud
ionativeformat)}/${SIPCHANINFO(t38passthrough)} QOS=${RTPAUDIOQOS})
The problems I have so far:
1. CODEC
Codec is reported only for A-Leg.
When transcoding asterisk logs the above line as: slin for read / slin for
write / the codec of A-Leg / 0 for t.38.
Is there a way to get the codec for both legs of a call?
2. RTP Qos is reported only for A-Leg.
Also, asterisk seems to ignore the RTP statistics reports by B-Leg after the
BYE:
-- Executing [...@core-dialplan:3] Hangup(SIP/401-08231540, ) in new
stack
== Spawn h extension (core-dialplan, h, 3) exited non-zero on
'SIP/401-08231540'
Scheduling destruction of SIP dialog
'0aa4f73f5c9715b7661b50080a669...@10.11.12.1' in 6656 ms (Method: INVITE)
set_destination: Parsing sip:4...@10.11.12.43:5060;transport=udp
sip:4...@10.11.12.43:5060;transport=udp for address/port to send to
set_destination: set destination to 10.11.12.43, port 5060
Reliably Transmitting (no NAT) to 10.11.12.43:5060:
BYE sip:4...@10.11.12.43:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport
From: Office Line 1 sip:4...@10.11.12.1
sip:4...@10.11.12.1;tag=as1d9352fe
To: sip:4...@10.11.12.43:5060;transport=udp
sip:4...@10.11.12.43:5060;transport=udp;tag=0009b7aa1aaa51eb2c767e13-7fb3b3
4a
Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1
CSeq: 103 BYE
User-Agent: home.tsaousis.gr
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (core-dialplan, 422, 1) exited non-zero on
'SIP/401-08231540'
box*CLI
--- SIP read from 10.11.12.43:50539 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.12.1:5060;branch=z9hG4bK3077b703;rport
From: Office Line 1 sip:4...@10.11.12.1
sip:4...@10.11.12.1;tag=as1d9352fe
To: sip:4...@10.11.12.43:5060;transport=udp
sip:4...@10.11.12.43:5060;transport=udp;tag=0009b7aa1aaa51eb2c767e13-7fb3b3
4a
Call-ID: 0aa4f73f5c9715b7661b50080a669...@10.11.12.1
Date: Sat, 21 Feb 2009 14:29:42 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0
RTP-RxStat: Dur=4,Pkt=180,Oct=28800,LatePkt=0,LostPkt=0,AvgJit=0
RTP-TxStat: Dur=4,Pkt=183,Oct=29280
These SIP messages are being exchanged after the dialplan has executed the h
extension.
Is there a way to have RTP statistics for both legs?
3. RTP IP is not reported anywhere.
The RIP= variable I have above, reports the SIP IP, and again only for
A-Leg.
Is it possible to find out the RTP (not SIP) IPs for both legs?
4. Find out who hangedup an answered call.
I have not found any way to determine the peer that requested to hangup the
call.
Is it possible to find who of the two legs requested the hangup?
Any help is appreciated.
Costa
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[asterisk-users] Script to softly restart Asterisk each midnight to clean locked channels
As Asterisk has inner problems and channels very often locks we have such
script to restart Asterisk each midnight.
We (our clients) mostly use v1.4.18.1. We can't upgrade to newer versions
because there are too much changes which would brake our system
(realtime/sip/iax2/cdr/etc/etc).
Script soft hangups all alive channels in dirty way then kills Asterisk and
starts it up.
Hope it will be useful to somebody.
Corrections/comments welcome.
#! /bin/sh
# Script to restart asterisk softly by Kolmisoft
# crontab
# 0 0 * * * /usr/local/mor/asterisk_nice_restart.sh
# tell Asterisk do not accept new calls
asterisk -rx 'stop gracefully' /dev/null
# read all channels
asterisk -rx 'core show channels verbose' | sed '1d' /tmp/f1
cat /tmp/f1 | awk '{split ($0,a, ); print a[11]}' /tmp/f2
# hangup all alive channels
for i in `cat /tmp/f2`; do
asterisk -rx soft hangup $i /dev/null
done
# let asterisk to stop by itself
sleep 5
# kill remainings
killall -9 safe_asterisk
killall -9 asterisk
# start fresh and ready to work!
/etc/init.d/asterisk start
# clean
rm -rf /tmp/f1
rm -rf /tmp/f2
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
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Re: [asterisk-users] Script to softly restart Asterisk each midnight to clean locked channels
Any guidelines how to solve locked channels problems? E.g. to find out which part of the code has problems and causes locks. Upgrade to newer versions are not an option. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: 2009 m. kovo 19 d. 17:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Script to softly restart Asterisk each midnight to clean locked channels Mindaugas Kezys escribió: As Asterisk has inner problems and channels very often locks we have such script to restart Asterisk each midnight. snip That is the things we must help to solve for not having to do to something like this on asterisk servers. Fortunately I use 1.4.22 version which has proved to me to be quite stable, judging from this uptime: System uptime: 1 week, 2 days, 19 hours, 22 minutes, 53 seconds Last reload: 10 hours, 31 minutes, 33 seconds Upgrading to 1.4.23.1 resulted in random core dumps (suspecting attended transfers issue) but unfortunately I've had no time to debug it and make a good bug report. My case is a 24/7/365 non-stop call center, so I didn't have another choice but to rollback. I hope some of us just can help asterisk be better by trying to use the latest version at least on testing environments, to not having to maintain an internal version and cherrypicking patches that may or may not resolve the issues that we could experience. Just my 2 cents... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- 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] Script to softly restart Asterisk each midnight to clean locked channels
Locked channel does not react to 'soft hangup' command.
That's why it is called - LOCKED.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Sent: 2009 m. kovo 19 d. 18:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Script to softly restart Asterisk each
midnight to clean locked channels
On Thu, 19 Mar 2009, Miguel Molina wrote:
Mindaugas Kezys escribi?:
As Asterisk has inner problems and channels very often locks we have
such script to restart Asterisk each midnight.
snip
Why restart Asterisk, free up the channel...
From cron, you can clear up any calls over say 3 hours:
/usr/sbin/asterisk -rx show channels concise|awk -F : '($11 10800)
{print /usr/sbin/asterisk -rx \soft hangup $1 \}'|sh
You don't necessarily have to keep restarting it at midnight.
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP
Enough research will tend to support your
conclusions. - Arthur Bloch
A conclusion is the place where you got
tired of thinking - Arthur Bloch
227C 5D35 7DCB 0893 95AA 4771 1DCE 1FD1 5CCD 6B5E
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Check this:
http://www.voip-info.org/wiki/index.php?page=Call+Quality+Metrics
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Sorry for replying for the second time, but this issue is interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this: http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
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---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com
Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Formula here: http://www.nessoft.com/kb/50 has jitter in it.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: Marc Leurent [mailto:lf...@leurent.eu]
Sent: 2009 m. balandžio 2 d. 13:56
To: asterisk-users@lists.digium.com
Cc: Mindaugas Kezys
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
Could you share with us your Openoffice callc function?
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc Leurent
Sent: 2009 m. balandžio 2 d. 11:29
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Extract a MOS value from Asterisk CDR
Indeed, we already have
- the function to convert R factor to MOS
- the R function R = R0 -Is-Id-Ie+A
- the codec used
- the rtt, rx/tx jitter, packet loss
What ye do not have but is needed:
- A factor, a note between 0 and 20 - 0 for landlines
- the Burst Ratio, I'm using 1 (random repartition)
I already have an openoffice calc function to calculate the MOS regarding the
rtt, packet loss, codec, I have to add the jitter!
Here are the URL I have used
* http://www.itu.int/rec/T-REC-G.107-200503-S/en
* http://www.ixiacom.com/library/white_papers/display?skey=voip_quality
* http://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
VTX, votre partenaire telecom proche de vous !
Le Thursday 02 April 2009 03.27:48 John Todd, vous avez écrit :
Thank you for the interesting links on MOS values and calculations!
It seems that many (most?) of the values that are used to construct R
and MOS could be obtained from the data that exists within the
dialplan, at least as far as the visible RTP path is concerned. Or
is there data missing in the current RTCP statistics that would be
required to make correct R/MOS value estimates? (If so, then that's
on-topic for asterisk-dev, otherwise this should be moved to asterisk-
users...)
Here is the data that I think is already visible:
- codec choices
- round-trip delay to RTP endpoint
- packet loss
- jitter
I think it is too complex to determine Irecency, A or packet loss
bursts unless there is significant additional code added to Asterisk
to capture more granular time-slices of data on each call. I also
think that mid-call codec changes should not be considered due to
complexity. Currently, I think this is un-necessary since most people
don't even seem to compute MOS to start with.
So in your examination you may come up with a script or dialplan that
creates a synthetic R or MOS value - could you post it to a blog, or
if it is very short, to the asterisk-users mailing list? I think this
would be worthwhile.
JT
On Apr 1, 2009, at 2:57 PM, Mindaugas Kezys wrote:
Sorry for replying for the second time, but this issue is
interesting for me
also.
I found such link: http://www.nessoft.com/kb/50
And this:
http://www.jdsu.com/product-literature/voipstats_an_acc_tm_ae.pdf
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marc
Leurent
Sent: 2009 m. balandžio 1 d. 18:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Extract a MOS value from Asterisk CDR
Hello all,
I'm tring to retrieve a formula to calculate a MOS value from
Asterisk RTCP
stats...
Have you got any idea how to do it?
Thanks
I'm reading all G.107 ITU docs to retrieve something...
I'm saving the SIP RTCP stats with:
[macro-hangupcall]
exten = s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to
calculate a MOS
value:
ssrc
=
592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.0
0;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
-- --
Marc LEURENT
lf...@leurent.eu
Re: [asterisk-users] error with dial timeout
Try this: Exten =_X.,n,Dial(SIP/ser_sei0/1130,,L(10208400:61000:1)) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: 2009 m. birželio 2 d. 11:07 To: asterisk-users@lists.digium.com Subject: [asterisk-users] error with dial timeout Hello, I am trying to do : Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Why? I forgot something ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- 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] SIP Strict Routing and canreinvite
Hello, I want to send Media outside Asterisk server, e.g. between peers. In CLI I see: . [Jun 8 13:13:58] VERBOSE[19112] logger.c: -- Native bridging SIP/5060-b7dc5218 and SIP/prov12-09ad3888 . [Jun 8 13:13:58] DEBUG[19112] chan_sip.c: Strict routing enforced for session 3ad367ee48778d2c523a60e62ae86...@85.113.41.129 And media still goes through Asterisk. Why is that? Why strict routing is enforced? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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] Sipura SPA-3000 vs Sangoma A200
Hello, Ok, few bad words about A200. Our company is based in Lithuania. Our company used SPA-3000, but because of echo problems we are not using them anymore. Now we are trying our luck with Sangoma A200 but the following problem occurred on few systems we installed. When calling person hangups the call, Asterisk (Sangoma) does not notice that and keeps ringing. Call is hangup only after 8s! That's totally unacceptable in busy call center. Imagine the agent which picks up the call and hears total silence. I tried all solutions found in Google. Most of them are on AsteriskGuru website. That didn't help. ringtimeout= in zapata.conf drives Asterisk crazy. Sangoma support (you will find it bellow) acknowledged that they don't know how to fix it on A200. Notice/question to Sangoma and others - why cheap SPA-3000 does not have this problem and more expensive A200 can't solve it? Can somebody suggets me working FXO alternative? Regards/Pagarbiai, Mindaugas Kezys Sangoma response to this problem -- Hi Mindaugas, Yuan asked me to respond because I have more experience in this. The way the phone system works is that when the phone is on hook, the line voltage is about 48V DC. To ring, the voltage increases to about 90v AC. When the phone or the FXO goes off hook, the line voltage is about 7 volts for the duration of the call. If the call is cleared at the far end, the voltage goes back to about 48 volts, and that tells us that the call has been terminated. On some systems they use a polarity reversal, or a 500ms drop of carrier current but the principle is the same. ON a good PSTN system, this change in voltage at the end of the call is almost instantaneous. In Canada, for instance, it takes over 10 seconds for the voltage signal to come through. The result is that on our Asterisk PBX at Sangoma, we have exactly the same problem as you: People call in, and hang up when they hear that you are not available, and we get messages about 10 seconds long with no audio. It is very annoying. We certainly would like to find a way around this ourselves. Bell Canada is not interested in our problems. We have tried using a silence filter to cut calls, but it happens often that there is a few seconds of silence in a call. Busydetect works, but the busy tone only comes much later, long after the 10 seconds has passed. I have no idea why some telcos have this delay before sending the disconnect signal. You may have better luck with your telco than we have had with ours. Please let me know if you find anything that helps. Regards, David Mandelstam Sangoma Technologies Corporation email: [EMAIL PROTECTED] web: www.sangoma.com Tel: 905-474-1990 x 106 800-388-2475 x 106 FAX: 905-474-9223 ___ --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] Sipura SPA-3000 vs Sangoma A200
Hello, Ok, few bad words about A200. Our company is based in Lithuania. Our company used SPA-3000, but because of echo problems we are not using them anymore. Now we are trying our luck with Sangoma A200 but the following problem occurred on few systems we installed. When calling person hangups the call, Asterisk (Sangoma) does not notice that and keeps ringing. Call is hangup only after 8s! That's totally unacceptable in busy call center. Imagine the agent which picks up the call and hears total silence. I tried all solutions found in Google. Most of them are on AsteriskGuru website. That didn't help. ringtimeout= in zapata.conf drives Asterisk crazy. Sangoma support (you will find it bellow) acknowledged that they don't know how to fix it on A200. Notice/question to Sangoma and others - why cheap SPA-3000 does not have this problem and more expensive A200 can't solve it? Can somebody suggets me working FXO alternative? Regards/Pagarbiai, Mindaugas Kezys Sangoma response to this problem -- Hi Mindaugas, Yuan asked me to respond because I have more experience in this. The way the phone system works is that when the phone is on hook, the line voltage is about 48V DC. To ring, the voltage increases to about 90v AC. When the phone or the FXO goes off hook, the line voltage is about 7 volts for the duration of the call. If the call is cleared at the far end, the voltage goes back to about 48 volts, and that tells us that the call has been terminated. On some systems they use a polarity reversal, or a 500ms drop of carrier current but the principle is the same. ON a good PSTN system, this change in voltage at the end of the call is almost instantaneous. In Canada, for instance, it takes over 10 seconds for the voltage signal to come through. The result is that on our Asterisk PBX at Sangoma, we have exactly the same problem as you: People call in, and hang up when they hear that you are not available, and we get messages about 10 seconds long with no audio. It is very annoying. We certainly would like to find a way around this ourselves. Bell Canada is not interested in our problems. We have tried using a silence filter to cut calls, but it happens often that there is a few seconds of silence in a call. Busydetect works, but the busy tone only comes much later, long after the 10 seconds has passed. I have no idea why some telcos have this delay before sending the disconnect signal. You may have better luck with your telco than we have had with ours. Please let me know if you find anything that helps. Regards, David Mandelstam Sangoma Technologies Corporation email: [EMAIL PROTECTED] web: www.sangoma.com Tel: 905-474-1990 x 106 800-388-2475 x 106 FAX: 905-474-9223 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOR and MCC - billing solutions for Asterisk released
Hello, Kolmisoft: http://www.kolmisoft.com released new versions of MCC and MOR - Billing solutions for Asterisk PBX MOR - is new product, it's MCC v2. Rewritten to support MySQL and is based on Ruby on Rails. ChangeLog for MCC: DB: New fields: providers.enable_cid_prefix varchar providers.disable_cid_prefix varchar providers.min_time integer default 1 providers.increment integer default 1 cids.enable_to_provider boolean default true cids.nat boolean default true cids.voicemail boolean defaut true cids.voicemail_psw charvar default '' rates.connection_fee double precision default 0 calls.rate double precision default 0 calls.user_connection_fee double precision default 0 calls.rate_connection_fee double precision default 0 users.first_name varchar users.last_name varchar users.min_time integer default 1 users.increment integer default 1 Fixed 386 code from Slovakia to Slovenia APP: Fixed bug with transfers - thanks German Aracil - suspended, needs more testing Changes to support CID manipulation - sponsored by Imre Csaba Varasdy Changes to support connection_fee based on rates(destinations) Rate, user_connection_fee, rate_connection_fee now are added to calls data Min_time and increment for billed time now taken from db, not conf file GUI: Fixed bug with email exists message Added Spanish translation - thanks German Aracil Added Hungarian translation - thanks Imre Csaba Varasdy Added German translation - thanks Inga A. Added Albanian translation - thanks Arben Myrtaj Now possible to assign connection_fee for rate(destination) User name split into First Name and Last Name Voicemail support in autoconfiguration, reachable by *98 for VoIP users User/admin can change cid/nat/voicemail/voicemail password for user's every CID (which supports autoconf.) under his details and when registering Possible to change call's status from processed to not (Changes color in GUI) and hide 'processed' calls in invoices. Possible to hide calls shorter than 'x' seconds. User can see his payments When registering, possible to set address like: http://mcc.company.com/register.php?ref=27, then referrer's field will be filled automatically Register authentication with noisy picture to prevent bot-registering Registration process reworked, check more here Reseller's CID's moved to new section - Devices Now various billing options (1/1, 6/6, 30/6) could be set per user basis - sponsored by Patrick Cardozo ASR (Average Success Rate) / ALOC (Average Length of Call) counting - sponsored by Patrick Cardozo New window to check CIDs and Extensions New values to define.php $USE_PROCESSED_CALLS $REG_ADDITIONAL - Additional info in registrtion page (like come visit our VoIP store) Regards, Midnaugas Kezys ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting sound before answer
Title: Message Hello, Lets say Im dialing out and before channels are bridged I hear beep or something similar. That way I know Im calling to other Telco/Provider. Is it possible to detect that beep before channel is answered and to redial through other trunk? Regards/Pagarbiai, Mindaugas Kezys ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Leave Queue when all agents busy
Hello, Does anybody knows how to make call to leave the queue when all agents in that queue are busy? Right now it tries to dial busy members and does not leave queue: -- Got SIP response 486 Busy Here back from 172.16.2.160 -- SIP/118-082252a8 is busy -- Called SIP/118 -- Got SIP response 486 Busy Here back from 172.16.2.160 -- SIP/118-082252a8 is busy -- Called SIP/118 -- Got SIP response 486 Busy Here back from 172.16.2.160 How to avoid that? I want to go to the next extension when all agents in the queue are busy. Thanks for help. Regards/Pagarbiai, Mindaugas Kezys ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MCC v.1.3 Released
MCC - Billing solution for Asterisk PBX Current version: 1.3 + 1.3.1 Patch MCC is a web-based, user (and admin) friendly billing interface for Asterisk and VOIP. MCC is open source software licensed under the GPL Some features of MCC: Unlimited SIP, IAX and Mobile/PSTN devices assigned to user Unlimited tariffs with different rates Rate Table viewable in Currency of choice Profit counting!!! Stats by countries Blocking of users Show Balance, Expenditure, Payments and number of Calls on each account Call Data Records (also in CSV/PDF) Advanced customer management and portal management Integrated PayPal and Hanza.net commerce modules View and Store Customers payments Manage Pre Paid and Post Paid customers. Full Credit control by User Account Concurrent calls for every user MCC Requirements: Asterisk PostgreSQL Apache + PHP Homepage: http://www.paskambink.lt/mcc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] morcdr v0.1 released
CDR Stats Analyzer and Report generator It's a rework of famous Asterisk Stats written by Areski. The main goal for this project is to concentrate more on PDF reports (managers love them!). Later more functions will be added. Please test it and send suggestions how to improve it. Licence: GPL Examples, demo and more info on homepage: http://www.paskambink.lt/mcc Regards, Mindaugas Kezys ___ --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] billing with PostgreSQL
You can try: http://www.paskambink.lt/mcc Regards/Pagarbiai, Mindaugas Kezys -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Wednesday, April 12, 2006 3:15 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] billing with PostgreSQL Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira ___ --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] the best billing tool for Asterisk
Hello, You can try: http://www.paskambink.lt/mcc Regards/Pagarbiai, Mindaugas Kezys -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Tuesday, April 11, 2006 9:55 AM To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] the best billing tool for Asterisk On 4/11/06 8:14 AM, Joao Pereira [EMAIL PROTECTED] wrote: Hello to all I would like to know some opinions of people that are using billing tools for Asterisk. Can you please advise me in wich billing tool to I use? Thanks Joao Pereira ___ --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 Lots of people whip together their own solution as there is no billing solution out there for Asterisk that fits all. Usually you end up making tweaks here and there even if you do use a prebuilt solution. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --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] Billing Server Open Source
http://www.paskambink.lt/mcc Regards/Pagarbiai, Mindaugas Kezys From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, April 17, 2006 8:13 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Billing Server Open Source Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MCC 1.4 released
Hello, New version of MCC Asterisk Billing Sofwate released. Many bugfixes and new functionality. Production ready version. Allready tested in 3 hardworking production environments! You can download preinstalled/preconfigured VMware image also. Everything here: http://www.paskambink.lt/mcc Regards/Pagarbiai, Mindaugas Kezys ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Applet to test VoIP quality
Hello, Does anybody know of free Java/ActiveX applet which could be placed into web and configured to check ping/latency/jitter/ports to selected server? Something like http://www.testyourvoip.com Regards/Pagarbiai, Mindaugas Kezys ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA2100 ringing without phone
Hello, We connected Sipura 2100 to Asterisk PBX. Plugged simple phone. Trying to call everything works ok. When we take out phone from Sipura, and trying to call, Asterisk shows, that Sipura is RINGING without phone connected to it. How could that be? -- SIP/240-2b03 is ringing How to tell SPA2100 to check for phone availability. Or is it just blind device which do not cares is phone connected to Line1 or not? Regards/Pagarbiai, Mindaugas Kezys ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling same queue member all the time
Hello, Im trying to setup a queue where call goes from agent to agent in strictly set order. I have queue (roundrobin): Agent1 penalty 1 Agent2 penalty 2 Agent3 penalty 3 When I call to this queue Agent1 rings. If this agent does not take the call, after set timeout same Agent1 is dialed again. The call never goes to Agent2 (only when Agent1 is not connected/paused and similar) So basically right now when Agent1 does not take the call dialing is: Agent1 Agent1 Agent1. I want: Agent1 Agent2 Agent3 What could be the problem? Regards/Pagarbiai, Mindaugas Kezys ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling same queue member all the time
Hello, Im trying to setup a queue where call goes from agent to agent in strictly set order. I have queue (roundrobin): Agent1 penalty 1 Agent2 penalty 2 Agent3 penalty 3 When I call to this queue Agent1 rings. If this agent does not take the call, after set timeout same Agent1 is dialed again. The call never goes to Agent2 (only when Agent1 is not connected/paused and similar) So basically right now when Agent1 does not take the call dialing is: Agent1 Agent1 Agent1. I want: Agent1 Agent2 Agent3 What could be the problem? Regards/Pagarbiai, Mindaugas Kezys ___ --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] Identifying invoking party for a feature
How did you managed to et info who pressed the button for feature request? CHANNEL and CALLERID(NUM) variables are pointing to wrong direction. Regards/Pagarbiai, Mindaugas Kezys -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne P. HIll Sent: Thursday, July 20, 2006 1:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Identifying invoking party for a feature heh. Just got permission to drop 1.2.10 on there and it seems to be working the way i want it now. --Wayne On Jul 19, 2006, at 3:40 PM, Wayne P. HIll wrote: I'm working on a server being implemented for a client right now which, due to a long string of issues I won't go into, has decided that they wish to use cisco 7960s over sccp with asterisk. Now it's up to us to write in the many features that this setup doesn't support by default. The current issue is with n-way calls. As a base we're using the code from the how-to on VoIP-info, using app_asyncgoto rather than using ChannelRedirect and running trunk. That, so far, hasn't been an issue. The How-to is here: http://www.voip-info.org/wiki/view/Asterisk+n- way+call+HOWTO for reference. The issue we face is this: Because of the way featuremap works, regardless of who invokes the feature, the party identified as 'caller' gets prompted to invite the 3rd party, eg Joe Random calls the clients company and it's determined a 3way conference is going to be necessary. Jenny Random, the Service Rep, hits the key sequence for a conference, with the current setup, Joe would be prompted for the party he wishes to conference, and Jenny would be thrown into the dynamic room. The questions I have is basically, is there any way to identify (through an accessible either as a variable or through agi) which party in a call is the party who actually invoked the feature? Thanks for the help Wayne ___ --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] Problem with AGI Script
Make sure /usr/bin/perl can be reached.
Also try in your CLI:
agi debug
Same case happens when I do not have php-cli installed for php AGI scripts.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, November 14, 2007 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem with AGI Script
I have asterisk 1.2.18 running on a new system we just installed. Although
I've used AGIs many times in the past, I'm stumped on this one. It may just
be a simple issue that I need another eyeset to look at.
My AGI does the following:
#!/usr/bin/perl
#Load a few modules...
use Asterisk::AGI;
use DBI;
$AGI = new Asterisk::AGI;
#Grab input from Asterisk
my %input = $AGI-ReadParse();
#Some Debugging
$AGI-exec('SayDigits',$ARGV[0]);
exit;
All seems fine. If I run the script from the command line it works as
expected:
[EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333
EXEC SayDigits 333
However, when actually running in practice I get:
-- Executing AGI(Zap/23-1, GetEmailfromDID.agi|5706016716) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
-- AGI Script GetEmailfromDID.agi completed, returning 0
extensions.conf
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,agi(GetEmailfromDID.agi|${CALLERID (number)})
exten = s,3,rxfax(${FAXFILE})
exten = s,104,Set([EMAIL PROTECTED])
exten = s,105,Goto(3)
Any thoughts on why asterisk doesn't seem to be passing anything to the
script and the script doesn't seem to be passing anything back? When I call
I do not hear the digits read to me, instead I just get thrown to the next
object after the digit reading.
___
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Re: [asterisk-users] pip tones in Monitor or MixMonitor
exten = _X.,1,Playback(beep) exten = _X.,2,MixMonitor. If you are starting the recording using some DTMF code sequence described in features.conf make sure you use caller, callee or both value to play sound to correct line end. Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Wednesday, November 14, 2007 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] pip tones in Monitor or MixMonitor Is there a way to enable the pip tones (beep) indicating that a call is being recorded? I know that ChanSpy does beep (unless q option is chosen) once, but not quite the same. ___ --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] Dialing time-out
You can always press # at the end of your number to send it to Asterisk. Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Thursday, November 15, 2007 6:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dialing time-out Ok, probably a dumb question. I believe I already I know the answer, but thought I would get feedback from others. One of the issues with user devices at the end Asterisk is dialing time out. This is a parameter within each hardware device. So if I set it to 3 seconds it appears from the moment after going off hook any key press starts a timer allowing me 3 seconds to enter the next number before Asterisk times out and generically says I'm am sorry that is not a valid extension. Now this is ok, of sorts. The fault in this is when you dial a valid number you are stuck waiting 3 seconds for the system to out pulse and connect. This clearly separates Asterisk from the traditional TDM platform behavior where a time out can be REAL LONG allowed people to dial at a snail's rate without upsetting the phone system but then immediately out pulsing when a number match is met, regardless if the number match is a 4 digit extension or 7 digit phone number. Is this one of the reasons and purposes Asterisk has a real-time option? Thanks, Jim ___ --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] pip tones in Monitor or MixMonitor
If you want peep every 15s, you should do:
[some_context]
exten = _X.,1,Set(LIMIT_WARNING_FILE=beep)
exten =
_X.,2,Dial(Local/mixmoncontext/#{EXTEN}||L(3600:3600:15000)\n)
[mixmoncontext]
exten = _X.,1,MixMonitor...
In [some_context] use L option variables:
* LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the
caller.
* LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee.
* LIMIT_TIMEOUT_FILE - File to play when time is up.
* LIMIT_CONNECT_FILE - File to play when call begins.
* LIMIT_WARNING_FILE - File to play as warning if 'y' is defined. If
LIMIT_WARNING_FILE is not defined, then the default behaviour is to announce
(You have [XX minutes] YY seconds).
More details about L option for Dial cmd:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack
Sent: Thursday, November 15, 2007 12:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] pip tones in Monitor or MixMonitor
I guess I didn't try this because Playback(beep) seems to me to playback the
beep once and not repeat ever 15 seconds as is needed for the pip tones.
Is this not true?
exten = _X.,1,Playback(beep)
exten = _X.,2,MixMonitor.
If you are starting the recording using some DTMF code sequence
described in features.conf make sure you use caller, callee or
both value to play sound to correct line end.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Plack Sent: Wednesday, November 14, 2007 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] pip tones in Monitor or MixMonitor
Is there a way to enable the pip tones (beep) indicating that a
call is being recorded?
I know that ChanSpy does beep (unless q option is chosen) once, but
not quite the same.
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Re: [asterisk-users] Best Prepaid Application?
You can try MOR FREE - it has nice gui and is very fast. LiveCD is available: http://www.kolmisoft.com/mor/content/view/83/95/ It is covered in extensive manual: http://www.kolmisoft.com/mor/component/option,com_remository/Itemid,40/func, fileinfo/id,25/ And yes - it's FREE as name suggests. Regards/Pagarbiai, Mindaugas Kezys Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT Sent: Friday, November 23, 2007 7:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Best Prepaid Application? Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch 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] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4
For testing purposes you can try one of these: http://kvin.lv/pub/Linux/Asterisk/ Mindaugas Kezys http://www.kolmisoft.com Advance Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fernando Berretta Sent: Friday, November 23, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4 Hi, I'm trying to install g729 codec in an Athlon 64 x2 Dual core processor 4000+ but.. all packages I've download haven't worked. Could someone please let me know what package should I download ? Best Regards, Fernando [EMAIL PROTECTED] modules]# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4222.52 processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 107 model name : AMD Athlon(tm) 64 X2 Dual Core Processor 4000+ stepping: 1 cpu MHz : 2109.624 cache size : 512 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm cr8legacy ts fid vid ttp tm stc [6] bogomips: 4219.18 ___ --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
Rename to codec_g729.so http://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] 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.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-athlon-sse.so codec_g729-ast14-gcc4-glibc-core2.so http://asterisk.hosting.lv/bin/codec_g729-ast14-gcc4-glibc-core2.so codec_g729-ast14-icc-glibc-x86_64-core2.so http://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
Re: [asterisk-users] Best Prepaid Application?
If you have any questions - there's forum on www.kolmisoft.com/mor to ask questions and get answers. Mindaugas Kezys http://www.kolmisoft.com Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT Sent: Monday, November 26, 2007 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Prepaid Application? Thank you for your answer! I'm going to try it! Have a nice day Mindaugas Kezys a écrit : You can try MOR FREE - it has nice gui and is very fast. LiveCD is available: http://www.kolmisoft.com/mor/content/view/83/95/ It is covered in extensive manual: http://www.kolmisoft.com/mor/component/option,com_remository/Itemid,40/func, fileinfo/id,25/ And yes - it's FREE as name suggests. Regards/Pagarbiai, Mindaugas Kezys Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT Sent: Friday, November 23, 2007 7:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Best Prepaid Application? Good evening, Have you got any idea which prepaid application will be the best to do simple prepaid calls with a MySQL storage...? PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch 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 ___ --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] To DB or not to DB?
Pros: 1. No need to reload Asterisk when you change settings 2. Changes are applied instantly 3. Easy to manage dialplan/users/settings 4. With properly programmed GUI you can give users some self-help services 5. No noticable overhead - dual xeon + 2gb ram does 400 simm. calls 6. You can have your DB on other server, that let's you connect several Asterisk servers to one DB - unified configuration Cons: 1. None Regards/Pagarbiai, Mindaugas Kezys Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Wednesday, November 28, 2007 6:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] To DB or not to DB? I lurk and comment a little on here and have been playing with * for a short while. I am interested in hearing about the pros and cons for using a database backend to Asterisk. My current setup is simple, out of the box with config files in /etc/asterisk and logs etc going into /var. I notice a great many of the contributors here seem to use a db backend (is this also called Real Time Asterisk?) and I'd like to know why if anyone cares to comment. Thanks Alan -- The way out is open! http://www.theopensourcerer.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] Answer Machine/Fax/modem detection
Maybe this can help: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tong Sent: Sunday, December 02, 2007 7:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Answer Machine/Fax/modem detection Has anyone sucessfully implimented a fax or modem detection dial plan? I'm originating calls from asterisk using a list of numbers and dropping the destination into an IVR menu but need to do something different if a modem or fax answers. I tried to use the NVBackgroundDetect() application but i think that is for receiving faxes only. Any help would be 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] Upgrade to Asterisk 1.4 - it's one year's old!
Hello everybody, Since 1.4 release our company installed more then 200 Asterisk servers using Asterisk 1.4 version. At start we had several bugs with SIP channel and CDR handling but starting from 1.4.6 or something it works without problems. We are really happy with 1.4 and thank you for your great job! Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E Sent: Saturday, December 15, 2007 12:57 PM To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] Realtime logic in Asterisk 1.4.16.1
Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=providers_IP and ipaddress=providers_IP Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/[EMAIL PROTECTED]) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' to retrieve info about this device. And in Asterisk 1.4.16.1 I see: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Where this came from? In mine DB host=providers_IP, how Asterisk managed to visualize that it should be dynamic?! Offcourse I get: [Dec 20 00:05:58] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:05:58] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --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 logic in Asterisk 1.4.16.1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Thursday, December 20, 2007 12:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On 00:12, Thu 20 Dec 07, Mindaugas Kezys wrote: Hello, I have configured one provider in Asterisk Realtime DB without username and password, only host=providers_IP and ipaddress=providers_IP Now when I'm trying to send call using this provider I'm using following string: Dial(SIP/[EMAIL PROTECTED]) In Asterisk 1.4.15 debug I see that Realtime engine is using query: [Dec 20 00:02:15] DEBUG[14634]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' to retrieve info about this device. And in Asterisk 1.4.16.1 I see: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Where this came from? In mine DB host=providers_IP, how Asterisk managed to visualize that it should be dynamic?! Offcourse I get: [Dec 20 00:05:58] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:05:58] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) Because Realtime Engine is not able to find my Provider which is NOT DYNAMIC! No other settings changed. Same configuration files. res_config_mysql.so recompiled to 1.4.16.1. Please help or explain what's wrong! Have a look at http://downloads.digium.com/pub/security/AST-2007-027.pdf That's why it's not working anymore -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? --- Thank you for pointing this, but I red this doc many times. It does not help. I tried to put username/password for my device - but it still is looking for dynamic. Does it mean I can't have anything else in host field for device except dynamic? Also this PDF states: An attacker may impersonate any user using host-based authentication without a secret, simply by guessing the username of that user. AFAIK host-based authentication is done by IP address. Username and password are not present. Following this I see no logic in above statements: host-based authentication without a secret - host-based auth. is always WITHOUT secret, and simply by guessing the username of that user - again - host-based auth. is always WITHOUT username If device (peer/user) has username/password - that's not HOST-BASED authentication. Correct me if I'm wrong. Question follows - how can I have host-based authentication in Realtime in Asterisk 1.4.16.1?? Maybe tommorow we will see Asterisk 1.4.16.2? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --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 logic in Asterisk 1.4.16.1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, December 20, 2007 1:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Correct, that's the FIRST lookup that is done. It then checks the IP address and does: SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89' where the IP address is what is sent in the SIP INVITE. If that fails, it does a lookup only on the name (old behavior). If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and checks every match for insecure=yes If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and checks every match for insecure=yes And if that fails, then it returns no match. So all of those queries had to run and fail for you to get no match. -- Tilghman -- Thank you for explanation, but problem is that only this first query is executed: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' [Dec 20 00:04:12] WARNING[25686]: chan_sip.c:2898 create_addr: No such host: Provider [Dec 20 00:04:12] WARNING[25686]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) That's it. No more queries. End of call. Why? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX ___ --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] ip phone suggestion for Asia?
Do not forget to evaluate Linksys SPA phones. Best I tried and not expensive. We use them (SPA942) in our company. Everybody's happy. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of d tbsky Sent: Thursday, December 20, 2007 6:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ip phone suggestion for Asia? Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --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] Active Calls
Happy Holidays! When call starts and your billing script/application starts – enter info about call in DB, like: datetime(call start), src, dst, RINGING, uniqueid When dialing: Dial(whatever|M(answer_mark_macro)) Macro: answer_mark_macro will put updated info to same row in DB: datetime(call answered), src, dst, ANSWERED, uniqueid When call ends your billing script/application should delete record from DB for this call. You can put TRANSFER info to DB also when transfer occurs. Also you can put any other info you find usefull about your call – codecs/phone model which is dialing and so on. This method lets you retrieve call status info from DB without using AMI – thus not bothering Asterisk at all. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR – Advanced Billing for Asterisk PBX From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abdul Sent: Monday, December 24, 2007 12:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Active Calls Hi Friends, Happy New Year I was developing billing system for my end user customers. I need to get Asterisk Active calls in MySQL database with full status of call likem ringing, UP and runtime? i will be thank full for your help and suggestion. Thank You _ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try http://us.rd.yahoo.com/evt=51733/*http:/mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ%20 it now. ___ --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 logic in Asterisk 1.4.16.1
Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 12:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 05:48:01 pm Mindaugas Kezys wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Thursday, December 20, 2007 1:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 16:12:27 Mindaugas Kezys wrote: [Dec 20 00:04:12] DEBUG[25686]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = 'Provider' AND host = 'dynamic' Note: host = 'dynamic' Correct, that's the FIRST lookup that is done. It then checks the IP address and does: SELECT * FROM devices WHERE name = 'Provider' AND host='23.45.67.89' where the IP address is what is sent in the SIP INVITE. If that fails, it does a lookup only on the name (old behavior). If that fails: SELECT * FROM devices WHERE host='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' AND port='5060' If that fails: SELECT * FROM devices WHERE host='23.45.67.89' and checks every match for insecure=yes If that fails: SELECT * FROM devices WHERE ipaddr='23.45.67.89' and checks every match for insecure=yes And if that fails, then it returns no match. So all of those queries had to run and fail for you to get no match. were you ever able to get a solution for this? i seem the same problem when storing my sip trunks in mysql, using 1.4.16.2 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --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 logic in Asterisk 1.4.16.1
Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right? And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user? I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot. Thanks for info. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E ___ --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 logic in Asterisk 1.4.16.1
Thank you! Will it come to 1.4.16.3 or 1.4.17? Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Sunday, December 30, 2007 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 06:30:09 Mindaugas Kezys wrote: Just want to double check. When you are using this for IAX2 then first query is with 'dynamic', right? And after that when no peer is found other query(-ies) are executed which retrieves correct info about IAX2 user? I will have to test this myself. If it is correct - then problem could be only for SIP and less trouble to troubleshoot. Thanks for info. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Messina Sent: Sunday, December 30, 2007 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Sunday 30 December 2007 05:34:12 am Mindaugas Kezys wrote: Currently we are using 1.4.15 which does not have such nasty BUG. When I will be free, I will try to review Asterisk sources to find a problem and submit patch to this. From this case I see that not much people are using Asterisk Realtime with newest Asterisk version. When holidays will end more and more people will start to complain about this. i found that it did not affect my iax2 tunks (outbound peers) in mysql realtime, but it did affect the sip trunks (outbound peers) in realtime. Please update to the latest SVN 1.4 -- this should have already been fixed. -- 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
[asterisk-users] Realtime device update weirdness
Hello,
We use Asterisk Realtime for our billing software. 200+ installations of
Asterisk with Realtime, but I see this for the first time.
Asterisk 1.4.17, Addons 1.4.5, No patches, no NAT - just plain simple
installation.
With debug I can see:
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:138 realtime_mysql: MySQL
RealTime: Retrieve SQL: SELECT * FROM devices WHERE name = '109' AND host =
'dynamic'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '0.0.0.0', port = '0',
regseconds = '0' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 0 rows on table: devices
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '0.0.0.0', port = '0',
regseconds = '0' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27885]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 0 rows on table: devices
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:662 mysql_reconnect: MySQL
RealTime: Everything is fine.
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:365 update_mysql: MySQL
RealTime: Update SQL: UPDATE devices SET ipaddr = '213.164.10.178', port =
'60854', regseconds = '1201750701' WHERE name = '109'
[Jan 30 22:38:21] DEBUG[27917]: res_config_mysql.c:379 update_mysql: MySQL
RealTime: Updated 1 rows on table: devices
Notice update: UPDATE devices SET ipaddr = '0.0.0.0', port = '0', regseconds =
'0' WHERE name = '109'
Correct behaviour is: UPDATE devices SET ipaddr = '213.164.10.178', port =
'60854', regseconds = '1201750701' WHERE name = '109'
Why update to 0.0.0.0 is executed? It makes devices unreachable. When device
reregisters - it becomes available for short time - then again - update to
0.0.0.0. Why it is happening?
For temporaly solution i had to patch res_config_mysql.c at line 342, added
such lines:
if ((!strcmp(newparam, ipaddr)) (!strcmp(buf, 0.0.0.0))){
ast_log(LOG_DEBUG,MySQL RealTime: Avoided to update %s to %s
!!!\n, newparam, buf);
ast_mutex_unlock(mysql_lock);
return -1;
}
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
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Re: [asterisk-users] G729 version to be downloaded for my machines
Download for Pentium4 Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Wednesday, January 30, 2008 10:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G729 version to be downloaded for my machines Hi List; The output of cat /proc/cpuinfo giving a [Intel (R) Pentium (R) D] so what is the g729 version I have to download to work with my machine? Any help? Regards Bilal 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] Asterisk's DANGEROUS Transfer CDR's
Hello, For such cases we usually suggest to put 2 boxes in your infrastructure: 1. Main billing gateway - where all PBX'es are connected (all client's remote PBX'es and your Local PBX) 2. Local PBX - where user's without PBX'es are connected Then user connects in following way: User - Local PBX - Main GTW - PSTN That way you will be save from transfer issue and all your clients will be able to transfer their calls on Local PBX. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man Sent: Wednesday, January 30, 2008 12:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's - Original Message From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 29 January, 2008 9:24:14 PM Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's The provider can disable transfers (which is what we do), but why can a PBX not still allow it? Our PBX customers all can do transferring... but that's because billing isn't needed THERE. The billing, if any, is done on our end, or their providers end. This really seems like a very small and moot point that is being blown up. Depends how much it could cost you I guess :). If you're not supporting transfers it's a moot point if you are it's a bit more interesting. If the receptionist needs to transfer the call, then she should be able to do that within the confines of her PBX... the transfer of her call should NEVER go back out her PBX back to the supplier, for if it does, her PBX now loses control of that call. Our customer base is residential and small business. They don't want to either pay for or support another a PBX thats what they've come to us for in the first place a lot of the time. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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?
My suggestion - use the distro which you know best. We use Debian (200+ installations). It works stable for us because we know how to achieve it. Others use Fedora/Centos - because they are experts in these systems. Stability and performance of the system does not depend on the distro - only on person who built this system. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of LWATCDR Sent: Friday, February 01, 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Enterprise or Fedora? We are using Fedora because that is what the company we got our system from recommended. If I was doing the system myself I would throw in my vote for CentOS. I am using it for a database server and I have had no problems with it at all. It is about as stable and secure of Linux distro as I have ever used. If you do go with them I suggest kicking the CentOS team a few dollars. They do dang good work. On Feb 1, 2008 9:16 AM, love U. all [EMAIL PROTECTED] wrote: i wanna build a production Asterisk box ,will RedHat Linux Enterprise Server be more stable than Fedora core Linux or it makes no significant difference Express yourself instantly with MSN Messenger! MSN Messenger ___ -- 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] Echo() app doesn't work
Hello, Seems you do not answer your channel before executing Echo(): -cut here--- Asterisk Ready. *CLI -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569 -- Accepting UNAUTHENTICATED call from 192.168.2.3: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing [EMAIL PROTECTED]:1] Verbose(IAX2/yassen-2, 1|Echo test application) in new stack Echo test application -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' -cut here--- Try this dialplan: exten = _X.,1,Playback(demo-echotest) exten = _X.,2,Echo() exten = _X.,3,Hangup or exten = _X.,1,Answer exten = _X.,2,Echo() exten = _X.,3,Hangup Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yassen Damyanov Sent: Saturday, February 02, 2008 10:08 AM To: Asterisk Users Mailing List Subject: Re: [asterisk-users] Echo() app doesn't work --- Tzafrir Cohen [EMAIL PROTECTED] wrote: -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' On which platform is that? Echo is executed, and exists without an error. Tzafrir, thank you very much for responding! Logs look the same everywhere (on all 32-bit platforms where Echo() doesn't work) and on the 64-bit xubuntu (where it does). The log says it exited non-zero, which does not seem normal to me, but nevertheless the log has that on the only working setup already mentioned. I guess it is not the platform but maybe some kernel stuff that breaks the thing... Please anyone, any hint? Thanks in advance! I paste here my original message for reference (no broken lines this time): -Original Message-- Date:Fri, 1 Feb 2008 17:01:56 -0800 (PST) From: Yassen Damyanov [EMAIL PROTECTED] Add to Address BookAdd to Address Book Add Mobile Alert Subject: Echo() app doesn't work To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed. Details: A. Platforms: -- AsteriskNOW 0.6 beta 32bit, updated; -- Debian Etch 32bit with stock kernel and native debian-packaged asterisk 1.2 -- Debian Etch 32bit with custom compiled kernel (Timer frequency 1000 Hz and couple more tweaks) and latest stable asterisk (1.4.17) compiled from source -- Ubuntiu 7.10 Server with native ubuntu packages (1.4.10) -- xUbuntu 7.10 Desktop on x86_64 (=amd64) with native ubuntu packages (1.4.10) Echo() works only on the 64-bit setup. Does not work for all other cases. The Playback() app works fine in *all* cases. (The microphone is tested and works fine, so it's not that simple!) For some of the setups I established two separate extensions and they could talk to each other (so important things work, yes). The logs show the same, that is, just what would be normal: -cut here--- Asterisk Ready. *CLI -- Registered IAX2 'yassen' (UNAUTHENTICATED) at 192.168.2.3:4569 -- Accepting UNAUTHENTICATED call from 192.168.2.3: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing [EMAIL PROTECTED]:1] Verbose(IAX2/yassen-2, 1|Echo test application) in new stack Echo test application -- Executing [EMAIL PROTECTED]:2] Echo(IAX2/yassen-2, ) in new stack == Spawn extension (phones, 500, 2) exited non-zero on 'IAX2/yassen-2' -- Hungup 'IAX2/yassen-2' -cut here--- My extentions.conf: -cut here--- [globals] [general] [default] exten = s,1,Verbose(1|Unrouted call handler) exten = s,n,Answer() exten = s,n,Wait(1) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() [outgoing_calls] [incoming_calls] [internal] exten = 500,1,Verbose(1|Echo test application) exten = 500,n,Echo() exten = 500,n,Hangup() exten = 501,1,Verbose(1|Playback test application) exten = 501,n,Playback(vm-review) exten = 501,n,Wait(1) exten = 501,n,Hangup() [phones] include = internal -cut here--- My iax.conf: -cut here--- [general] bandwidth
Re: [asterisk-users] Best ATA. Period.
Linksys SPA 2102. No issues at all. Period. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, February 20, 2008 11:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Best ATA. Period. Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works? ___ -- 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 GSM
Cyber-Telecom's CT-V372 is same box as PorTech MV-372 but with more advanced firmware. It supports more functions, such as SMS sending. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Willcox Sent: Wednesday, February 20, 2008 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP GSM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Tuesday, January 29, 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP GSM With that sort of set up, If for example i get a 8 channel GSM gateway and the X100P can i make more than 1 concurrent call though the gateway with the X100P or does it only support 1 call at a time? What im looking to do is get a multi channel GSM gateway, and have the ability to make more than 1 call at once through it. The PorTech MV-372 works nicely with asterisk and is multichannel (2, if that counts!) Cheers, Ben ___ -- 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] chan_h323 requirements
This can help (script for Debian): apt-get install flex bison #dirty hack to prevent error from missing file cd /usr/include/linux touch compiler.h #PWLIB cd /usr/src wget http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz tar zxvf pwlib-v1_10_0-src-tar.gz cd pwlib_v1_10_0/ ./configure make make install make opt PWLIBDIR=/usr/src/pwlib_v1_10_0 export PWLIBDIR #OpenH323 cd /usr/src wget http://ovh.dl.sourceforge.net/sourceforge/openh323/openh323-v1_18_0-src-tar.gz tar zxvf openh323-v1_18_0-src-tar.gz cd openh323_v1_18_0/ ./configure make make opt make install OPENH323DIR=/usr/src/openh323_v1_18_0/ export OPENH323DIR cd /usr/src/asterisk/channels/h323/ make make opt cd /usr/src/asterisk ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig #or similar way #cp /usr/local/lib/* /usr/lib Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, February 21, 2008 10:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_h323 requirements Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- 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] How to get a clean, basic configuration?
We do: in modules.conf: noload = pbx_ael.so noload = pbx_dundi.so noload = res_config_pgsql.so noload = res_smdi.so in extensions.conf delete every context [default], [demo], whatever in sip.conf, iax.conf delete all peer/users if any Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vincent Sent: Thursday, February 21, 2008 4:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to get a clean, basic configuration? Hello I'm using a standard Asterisk install with default settings, and when I run reload, I see that Asterisk fetches configuration information from a lot more sources than just my extensions.conf and sip.conf. For instance: -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to 'us' == Parsing '/etc/asterisk/features.conf': Found == Parsing '/etc/asterisk/adsi.conf': Found == Parsing '/etc/asterisk/dundi.conf': Found == Parsing '/etc/asterisk/extensions.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4094 pbx_load_module: Starting AEL load process. [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4101 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. etc. How can I go and trim things down? Thank you. ___ -- 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] Voted most stable and easy to use phone?
Linksys SPA942. Tried most of available phones on the market. These phones sits on companies tables for more then a year. No problem at all, easy to use, nice(!) to use. I recommend to everybody. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, February 22, 2008 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voted most stable and easy to use phone? On Thu, Feb 21, 2008 at 7:32 PM, arkda [EMAIL PROTECTED] wrote: I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are pretty straightforward to manage via TFTP, and work really well with Asterisk. I agree, we've had zero trouble with these. Easy to install and they just work. ___ -- 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] mysql sip realtime
If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime peername from the CLI Without reload prune does not take effect in 1.4.x And after reload all registrations are lost. So basically Asterisk Realtime is big mess from our experience and is totally unusable. We ended making #exec based script which takes data from DB and forms static configuration on each reload. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 2009 m. rugpjūčio 20 d. 15:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mysql sip realtime Hi The column order in your mysql sip table is irrelevant (Example sip table here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip) All generic parameters are still taken from sip.conf and you must set rtcachefriends=yes If you change anything in your mysql sip table you do not need to reload the modue, what you need to do is sip prune realtime peername from the CLI As stated previously, you should never have to reload the sip module once realtime is working properly Hope this all helps Ish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
