Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0.
Hi guys, I need a desperate help from you regarding this asterisk crash issue. On Thu, Feb 13, 2014 at 5:48 PM, Arun Ram arunram@gmail.com wrote: Hi, I am facing asterisk crash issue in my Asterisk 10.0.0. safe asteriskgenerated a core dump in /tmp path . I viewed the core dump using viewcore in linux. *can anyone tell the reason for the crash . waiting eagerly for an answer from asterisk support guys*.* please the find the core dump attachment too* .. *Below is the information in core dump * -- *Thanks RegardsArunram.c* *The Power of someone has the power to do something.. anything !!* -- *Thanks RegardsArunram.c* *The Power of someone has the power to do something.. anything !!* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0.
Hi Eric Wieling, Thanks for your reply. what is the reason for that crash?? . when i read the core dump i found something like signal 11. what it means because of signal 11 asterisk crashed . Before upgrading i need to submit a report to my team for that i need a valid reason for that crash. .: NOTES INFORMATION :. ### found note section at offset: 0x4294 ### --- note 0 at offset 0x4294 --- padding: 4 bytes note name size: 0x5 bytes note description size: 0x90 bytes note name: CORE note type: PRSTATUS [1] signal number: 11 extra code: 0 errno: 0 *current signal: 11* set of pending signals: 0 set of held signals:0 pid:5136 ppid: 5136 pgrp: 2770 sid:2101 user time: 0.32994 sec system time:0.26995 sec cumulative user time: 0.0 sec cumulative system time: 0.0 sec bool pr_fpvalid:1 On Fri, Feb 14, 2014 at 10:57 AM, Eric Wieling ewiel...@nyigc.com wrote: Upgrade to 11. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Friday, February 14, 2014 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0. Enable debugging module and backtrace and re-compile so that you will bactrace of the crash logs. Regards On 14 Feb 2014 10:29, Arun Ram arunram@gmail.com wrote: Hi guys, I need a desperate help from you regarding this asterisk crash issue. On Thu, Feb 13, 2014 at 5:48 PM, Arun Ram arunram@gmail.com wrote: Hi, I am facing asterisk crash issue in my Asterisk 10.0.0. safe asterisk generated a core dump in /tmp path . I viewed the core dump using viewcore in linux. can anyone tell the reason for the crash . waiting eagerly for an answer from asterisk support guys. please the find the core dump attachment too .. Below is the information in core dump -- Thanks Regards Arunram.c The Power of someone has the power to do something.. anything !! -- Thanks Regards Arunram.c The Power of someone has the power to do something.. anything !! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Thanks RegardsArunram.c* *The Power of someone has the power to do something.. anything !!* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax issue
Hi, I am using asterisk 1.8.9.3 , I am using Spandsp FAX Driver: 20091228 123351, i will clearly explain my scenario. I am sending fax to a anlog fax machine which is connected to mediatrix analog gateway. say the analog fax extension is 18260. I have enabled the fax mode , T.38 codec, clear channel codec , cng tone everything in my analog gateway. while sending a fax from asterisk to the analog fax extension(18260 this extension is registered in asterisk) , I am using SendFax dialplan application for sending fax , most of the time i got success message with fax send status success and my anlog fax machine also receives the fax and printing it. but some time i am getting a error that coud not generate CNG tone in channel sip/18260. i saw this error in asterisk CLI. i want to know whether this issue in my asterisk side or in my gateway side. why asterisk not able to generate the CNG tone on this channel. second thing is in my sip.conf for fax I have enabled t38pt_udptl = yes,fec,maxdatagram=400 faxdetect = yes I want to know which error correction i can use whether FEC or Redundancy so that i can get maximun throughput while sending fax from my asterisk In my res_fax.conf maxrate=14400 minrate=2400 statusevents=yes modems=v17,v27,v29 ecm=yes I need a desperate support from this forum to solve this issue (coud not generate CNG tone in channel ). -- *Thanks Regards Arunram.c * *The Power of someone has the power to do something.. anything !!* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy landysacco...@yahoo.comwrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manual Web-meetme
how about this http://www.voip-info.org/wiki/view/MeetMe-Web-Control On Sat, May 22, 2010 at 9:45 PM, Renato bianchini renato...@yahoo.com.brwrote: Hi anyone, I need to install an application to organize conference in Asterisk, and to this I wanna use webmeetme, but I don't get find a good manual, anyone have or know where I can find a good manual to this application? Thank you very much. Renato -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voipmonitor.org
On Mon, May 10, 2010 at 1:09 PM, Martin Vít v...@lam.cz wrote: On 8.5.2010 00:40, Jeff Brower wrote: Martin- checkout new open source voipmonitor.org SIP packet sniffer. I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution. Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer. How many channels can it handle simultaneously? I've not tested limits but capturing 15 voip calls takes 3-4% on Core2 2.40GHz. Complexity in worst case is O(N^2) where N is number of calls. Packets are matched as llinear list of IP and port. If this will be limit, it could be rewriten to hash table O(N) How does it do MOS prediction if low bitrate codecs are being used (G729, AMR, etc)? It is calibrated only to G.711 with PLC for now but I'm planing adding equations for G.729 and iLBC. when are you expecting to release Ram -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re TrixBox
its possible ask the same question trixbox forum Ram On Sun, May 9, 2010 at 6:45 PM, Samantha saman...@femtech.com.au wrote: Hey Guys We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the customer wants to move the callcentre. They are asking for an equiv to the ipview I gather HUD may be or the panel view The problem is that we need to see (a) total calls in the queue (b) calls for specific DID - How can you give 1 DID preference to another DID ie DID 61740410001 = Fred Electrical DID 61740410002 = Bus Tour ABC DID 61740410003 = Fred 24/7 Plumber SO in A we need to see how many calls waiting So in B we need to see how many calls are waiting for DID 001 002 and 003 finally if there are 20 calls on hold in DID 001 and 002 and there is a call on 003, how can we place that to the top of the queue? thanks Samantha -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kamailio
Hi what do you want to integrate, Media Services or Loadbalance ? Ram On Mon, Apr 19, 2010 at 4:14 AM, Hector Muñoz hectormun...@gmail.comwrote: Hi guys, I want to integrate with two asterisk servers a kamailio sip server. Any of you know some good tutorial for this? Thanks in advance! Regards. -- jabber: trip...@12jabber.com blog: http://impresionesdeunloco.wordpress.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple instances of asterisk on same machine
On Wed, Dec 30, 2009 at 5:29 PM, Saeed Akhtar saeedakhtar@gmail.comwrote: hi all, I have a little problem I'm using asterisk with opensips as opensips dispatches calls to asterisk. I have to use multiple asterisk servers but since for the time being im using 1 machine for testing i want to run different instances of asterisk running on 1 pc listening to different ports. Can someone please guide me how to do this? I'll be very thankful how about configuring different config files in different folders and run asterisk to use that config files. ram ___ -- 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] error when open a2billing web page!
On Mon, Dec 28, 2009 at 10:39 PM, Zhang Shukun bit...@gmail.com wrote: hi, i have installed a2billing , when i open /admin web pages. errors as follow: Fatal error: Call to undefined function bindtextdomain() in /usr/local/src/a2billing/common/lib/languageSettings.php on line 130 do you know what's wrong? you get quick responce if you post the same in a2bill forum look at their site forum Ram ___ -- 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] Opensips+asterisk problem
Authentication Required. Via: SIP/2.0/UDP ipphone:5060;received=ipphone;branch=z9hG4bK1984515716453028636;rport=5060. From: 4720779942 sip:4720779...@asterisk-a2b-ip:5060;tag=3037030266. To: 0017 sip:0017x...@asterisk-a2b-ip:5060;tag=as0cb075c5. Call-ID: 14399316162240-7371067914...@ipphone. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=07ba8624. Content-Length: 0. -- when i enable debug at Asterisk and Look at i see the below error --- --- SIP read from a2b-asterisk-ip:5060 --- INVITE sip:0017xx...@a2b-asterisk-ip:5060 SIP/2.0 Record-Route: sip:a2b-asterisk-ip;lr=on Via: SIP/2.0/UDP a2b-asterisk-ip;branch=z9hG4bK166.1b7e2827.0 Via: SIP/2.0/UDP Ip-phone:5060;received=Ip-phone;branch=z9hG4bK295731884823024293;rport=5060 From: 4720779942 sip:4720779...@a2b-asterisk-ip:5060;tag=12544334 To: 0017X sip:0017xx...@a2b-asterisk-ip:5060 Call-ID: 16946271051109-143302828620...@ip-phone CSeq: 1 INVITE Contact: sip:4720779...@ip-phone:5060 Max-Forwards: 69 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 319 v=0 o=4720779942 31008195 22123120 IN IP4 Ip-phone s=A conversation c=IN IP4 Ip-phone t=0 0 m=audio 10030 RTP/AVP 18 4 8 0 9 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv - [Jun 30 01:15:29] VERBOSE[24612] logger.c: --- (15 headers 14 lines) --- [Jun 30 01:15:29] VERBOSE[24612] logger.c: Ignoring this INVITE request [Jun 30 01:15:31] VERBOSE[24612] logger.c: Reliably Transmitting (no NAT) to termination-provider-ip:5062: OPTIONS sip:termination-provider-ip:5062 SIP/2.0 Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport From: asterisk sip:aster...@a2b-asterisk-ip:5062;tag=as4cf91fd8 To: sip:termination-provider-ip:5062 Contact: sip:aster...@a2b-asterisk-ip:5062 Call-ID: 65a49c0977c6de0a1d2dbbfe75772...@a2b-asterisk-ip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jun 2009 08:15:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 30 01:15:32] VERBOSE[24612] logger.c: --- SIP read from termination-provider-ip:5062 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport=5062 From: asterisk sip:aster...@a2b-asterisk-ip:5062;tag=as4cf91fd8 To: sip:termination-provider-ip:5062;tag=2560d490c3265ff35995c6bbde62a7c3.ee5a Call-ID: 65a49c0977c6de0a1d2dbbfe75772...@a2b-asterisk-ip CSeq: 102 OPTIONS Content-Length: 0 - why does Asterisk sending with out any values --- From: asterisk sip:aster...@a2b-asterisk-ip:5062;tag=as4cf91fd8 To: sip:termination-provider-ip:5062 --- Any suggestions Ram ___ -- 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] Vicidialnow
On 1/23/09, David @ULC ucoms2...@gmail.com wrote: But after installing it with CD , I guess we have to change SIP file and do few more changes .. I am looking for those steps.. On Fri, Jan 23, 2009 at 2:55 PM, David @ULC ucoms2...@gmail.com wrote: Anyone have properly formatted document ? On Fri, Jan 23, 2009 at 1:42 AM, David @ULC ucoms2...@gmail.com wrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.comwrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? Buy manuals and understand the Dialplan logic Ram ___ -- 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] Vicidialnow
On 1/22/09, David @ULC ucoms2...@gmail.com wrote: But I believe even after doing that , there are few setting and changes required before we can start using it for production I guess... On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.com wrote: Anyone using VicidialNow ? I have documents for Vicidial scratch install but how to install step by step Vicidialnow ? its working for several people over net.. Ram ___ -- 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] * + Legacy PBX works but strange problem
On Sun, Nov 16, 2008 at 9:41 PM, Sriram [EMAIL PROTECTED] wrote: hi Robert followed your points - but problem persists...everything goes well for sometime but after that - asterisk is unable to dial the pbx... any more thoughts Post some outputs or logs ram ___ -- 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 bind a SIP channel to an IP
On Mon, Oct 27, 2008 at 7:12 PM, srinivas Antarvedi [EMAIL PROTECTED] wrote: Hello members, Mysetup: Asterisk 1.4 Phones:Polycom501 I wanted to register my polycom phones only from a fixed IP(on LAN ) i tried following scenarios and my results are described as follows 1)sip.conf [xxx] host=192.168.0.15 result is after some time the registration expires and i was unable to receive calls on my channel... 2)sip.conf [xxx] defaultip=192.168.0.15 i) result is after some time the registration expires and i was unable to receive calls on my channel ii)it is even allowing me to register from another ip address say 192.168.0.16 3)sip.conf [xxx] host=dynamic defaultip=192.168.0.15 in this case i dont have any problems and it was working fine... can anybody helpme out to bind the phones to a particular ip if not is it possible to do at all just give me a hint so that i will work on Look out some examples here http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf ram ___ -- 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] openser+asterisk
On Mon, Oct 27, 2008 at 11:53 AM, jordan pan [EMAIL PROTECTED] wrote: Hi everyone, I want to use the openser and asterisk to create a system ,who can give me a detail example about it,i found it have some complicated. Thanks in advance. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg60425.html Look the above link for your requirement Ram ___ -- 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] SER + Asterisk
On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote: On 10/17/08 23:23, Kristian Kielhofner wrote: On 10/17/08, Alex Balashov [EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. I would gladly go with any of the newer packages if I only could. I'm just working with what I can find in portage; I'm sure it will be eventually available. It will first show up via overlay. What I'm trying to do is to register SER to my VoIP provider via stun.fwdnet.net and connect SER with Asterisk, I just need some simple practical example; and upgrade will come with time. I'm sure it is possible even with old SER. Suggesting what is newer is not going to help me much :-) Hi Joseph you can use UAC Module to register with provider and make calls using SER/Openser/OpensSIPs or you can do other way is SER as registrar and Asterisk act a b2bua ( you can register with provider) let me know if it helps your need Ram ___ -- 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] prective dialer
look at Vicidial ram On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim [EMAIL PROTECTED] wrote: hi everybody This is Yavuz YILDIRIM I am software developer.I have a some problems in asterisk. I am using mysql db. Realtime using asterisk modules. On db i am using calling hundred fields for use dial. But i don't know how i can automaticly dial this fields on records numbers. Who can help me asterisk api and others. 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] Vividial issue
On Sun, Sep 28, 2008 at 8:16 AM, Brad [EMAIL PROTECTED] wrote: does anyone have a sample dialplan for vici dial that does not include any pri stuff. I am running exclusively SIP for everything and trying to edit the sample dialplan and removing anything to do with a pri card is becoming a nightmare! Thank you! check in the source there are lot of sample configs shown in the SVN Tree ram ___ -- 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 User with Calling Card Question
On Sun, Sep 28, 2008 at 3:56 AM, Babcock, Michael Alex [EMAIL PROTECTED]wrote: can a2 billing work on the same system that directadmin is installed? should not be a problem ram ___ -- 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] Setting up Asterisk to make calls using a VoIP provider and the regular phone line
Hi Anotherthing you need to consider regulations in india Calling from Outside world to India PSTN Using and VOIP PBX is Illegal. Only allowed right now Calling From IP Phone to any part of the world see the regulation site dotindia.in for more information ram On Sat, Sep 6, 2008 at 9:26 PM, logan [EMAIL PROTECTED] wrote: Hello Everyone, Thanks for the answers, guys. DID won't work for me as I'm more interested in making calls using the line in India. Trixbox it's going to be my installation and I went through the recommended hardware on there site and I'm thinking of getting a Polycom 330. But can anyone tell me if there is a cheaper phone for me to use and which is well compatible with Trixbox (Polycom 330 is about $120 for me and something around $60-70 will be great)? I don't want all the fancy features, just something plain and simple to use. Coming to the FXO cards, I'm considering for Linksys SPA3102NA (successor of sipura 3000). I just want a second opinion from you guys if it's a good choice or there are better and cheaper options out there. Thanks a lot everyone. Best Regards, Hitesh - Original Message - From: Nhadie [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, September 20, 2008 1:09 AM Subject: Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line Hi Hitesh, Usually, subscribing to DID provider is a one way thing, they can call you to that number, but you cannot call out via that number. If you already have a pots line available, which means you are probably paying monthly for it already, might as well buy an fxo card and make use of the line. anyone in india can call you locally and you can call anyone in india using the same line, as everyone is suggesting, use trixbox, not much linux experience is required, just boot from the cd and let it install itself. you might need linux experience when you compile drivers for your fxo card though, but they usually come with instructions which is quite easy to follow. hth regards, nhadie logan wrote: Hi Jai, If I understand correctly then the DID will enable to call me on the hardphone connected to the Asterisk. Will it also enable me to call out using the PSTN line at my home in India from Canada? Thanks. Best REgards, Hitesh On Fri, Sep 19, 2008 at 10:33 AM, Jai Rangi [EMAIL PROTECTED] wrote: Hitesh, If you dont have experience with Linux I would recommend you to use Trixbox, that will come with all the required packages and will do everythign for you. Re: FXO and FXS, you don't need to buy any card for True VoIP. Now you can buy DIDs that can come to your asterisk over the internet. Jai www.didforsale.com *Buy SIP DIDs at low cost unlimited minutes http://www.didforsale.com; On Fri, Sep 19, 2008 at 9:18 AM, logan [EMAIL PROTECTED] wrote: Hello Ram, Thanks for the response. As I said there are too many options out there :). Could you help me in settling down on one? Something that will work with the phone lines in India is just fine for me. I don't have any or much Linux experience, but willing to play around, so any compatible distro will do for me. So once again: Which Linux distro is best with Asterisk? Which hardphone is the easiest to setup? Which fxo/fxs card I should go for? Thanks a lot guys. Best Regards, Hitesh On Thu, Sep 18, 2008 at 10:33 PM, ram [EMAIL PROTECTED] wrote: On Wed, Sep 17, 2008 at 1:10 PM, logan [EMAIL PROTECTED] wrote: Thanks a lot Nhadie. I appreciate your help. Could you also suggest some brands or models of the FXO+FXS card that are seamlessly compatible to Asterisk? Also what hardphone I should go for as there are so many in the market? What should be the configuration of the system running this kind of Asterisk setup? And which Linux distribution is best suited with Asterisk? Hi you can look this compatable hardware http://www.voip-info.org/wiki/ http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems http://www.voip-info.org/wiki/view/VOIP+Phones Its very difficult to say which OS is good, its all depends on your experience and your hands on the same. Look at Trixbox, its automated CD ram ___ -- 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
Re: [asterisk-users] Setting up Asterisk to make calls using a VoIP provider and the regular phone line
On Wed, Sep 17, 2008 at 1:10 PM, logan [EMAIL PROTECTED] wrote: Thanks a lot Nhadie. I appreciate your help. Could you also suggest some brands or models of the FXO+FXS card that are seamlessly compatible to Asterisk? Also what hardphone I should go for as there are so many in the market? What should be the configuration of the system running this kind of Asterisk setup? And which Linux distribution is best suited with Asterisk? Hi you can look this compatable hardware http://www.voip-info.org/wiki/ http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems http://www.voip-info.org/wiki/view/VOIP+Phones Its very difficult to say which OS is good, its all depends on your experience and your hands on the same. Look at Trixbox, its automated CD ram ___ -- 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] Dialplan Help
On Thu, Mar 20, 2008 at 8:37 PM, Jeremy Mann [EMAIL PROTECTED] wrote:
I've got a couple of extensions in users.conf that have both SIP and IAX
access(IAX softphone, SIP hard phone).
I'd like to setup my dial string to check to see which they are actively
registered with, and send the call appropriately.
Right now I have:
Exten = _4xx,1,Dial(SIP/${EXTEN}IAX2/${EXTEN})
But not all phones have both techs, so there is a lot of misses
Is there a way to use the hints to see which they are registered with, and
dial only using those channel types?
you can mention the context
and dial plan so that respective users go to the same channels
they belong to
ram
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Re: [asterisk-users] TDM800P FXO problem incomming call
On Jan 22, 2008 1:50 PM, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk 1.4.11 on Cent 4.3 i have faceing some problem i have TDM800P 8 port FXO card when i terminate PSTN line on this port can make outgoing call it is working fine but incomming call not handling ...when i call from outside to this line it is rinning but no one call land on my asterisk no debug in asterisk some time it land but most of time not . check the dialplan to match or contact provider for the problem. simple solution..take the line connect to phone ( if not e1), check incoming call coming or not ? ram ___ -- 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 optimalization
http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm check this link may help you ram On Jan 23, 2008 10:23 PM, marek cervenka [EMAIL PROTECTED] wrote: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- 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 block spammer calls
Hi I am setting up a Calling card Plat form I have incoming toll number, the provider charges incoming calls I see some spammers( competetors) keep calling my toll. so iam getting huge invoices how can i identify those kind of spammers and block the callerID for some time any suggestions or example could help me ram ___ --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 as sip server
On Jan 5, 2008 4:40 AM, ameel [EMAIL PROTECTED] wrote: I am trying to setup asterisk as a registrar and sip server only. Currently When I make calls all my rtp traffic is going through the asterisk server as a B2BUA. Is it possible to turn off this feature and have all my calls RTP traffic going directly to the SIP UA? __ Hi Try SER or OpenSEr ram ___ --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 block spammer calls
On Jan 5, 2008 11:26 AM, Trevor Peirce [EMAIL PROTECTED] wrote: ram wrote: Hi I am setting up a Calling card Plat form I have incoming toll number, the provider charges incoming calls I see some spammers( competetors) keep calling my toll. so iam getting huge invoices how can i identify those kind of spammers and block the callerID for some time any suggestions or example could help me If the caller doesn't key in a valid PIN after so many tries, disconnect them. If they are disconnected more than so many times, block them. If you'd like to hire someone to implement this for you, you'd be better served by posting to the -biz list and asking for assistance there. Hi I understand what you are saying. so once we see he is not input the pin more than 2times he will be blocked for hour ( i will run cron job, after one hour release them) is this a good idea. ram ___ --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] Didnt get a frame from Channel and call gets disconnected
On Dec 11, 2007 11:56 PM, Jai Rangi [EMAIL PROTECTED] wrote: Anyone, could you please suggest the latest stable release for asterisk. -Jai on 1.2 1.1.24 or try latest SVN 1.4 ram ___ --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] Enable/Disable Sip without registration
On Dec 12, 2007 6:31 PM, equis software [EMAIL PROTECTED] wrote: I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn´t found wath opition configure this functionality. Create a users in sip.conf with context so that user will register with asterisk to make calls ram ___ --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 setup redundant SIP peers
On Nov 30, 2007 11:46 PM, Thomas Balsfulland [EMAIL PROTECTED] wrote:
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten = _0Z.,1,Dial(SIP/49${EXTEN:[EMAIL PROTECTED],30)
add another line
exten = _0Z.,2,Dial(SIP/49${EXTEN:[EMAIL PROTECTED],30)
ram
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Re: [asterisk-users] Softswitch digim
On Dec 3, 2007 3:12 AM, Carlos Rojas [EMAIL PROTECTED] wrote: Hello averybody, I'm looking the softswitch in digium website, anyone test the softswitch? Try freeswitch.org ram ___ --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] Newb Question
chan spy does the job i belive ram On Nov 30, 2007 7:37 AM, Jeff Adams [EMAIL PROTECTED] wrote: I inherited an office with phones that are hosted off-site. Everything is skinny and G729. I see that the FreeBSD asterisk port comes with a G729 codec. I want to record everything. If I use port mirroring on my switch, is it possible to configure asterisk to record and assemble packets that it doesn't otherwise route? Is it insane to user asterisk for this purpose? Advice or a link to a howto would be greatly appreciated. ___ --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 Help
On Nov 7, 2007 12:55 AM, Jarga Jallow [EMAIL PROTECTED] wrote: Under asterisk info: Sip registry12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? Thanks in advance. qulify=yes or time in seconds should solve this issue ram ___ --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 and Grandstream both behind different NAT
On 11/6/07, Vivek Shrivastava [EMAIL PROTECTED] wrote: Hi, i am facing some problem configuring 2 Grandstrem phones 101 behind NAT. I have put server ip as 192.x.x.x and OutGoing proxy as 72.x.x.x, i have forwarded ports on both Grandstream and Asterisk sides, and using those ports on Grandstream for SIP and RTP with random ports =no. This setup is working however at a time only one phone gets registered. Has someone experienced the same problemany suggestions? use ngrep to do network trace ram ___ --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] Linksys SPA-941 Unavailable
On 11/6/07, Kim Joung-il [EMAIL PROTECTED] wrote: Hello! We are using several Linksys SPA-941 in our office. After IP change occur devices seems not to be reachable, actually unavailable! Devices is connected, e.g. we can place a call using SPA-941 but can not receive any calls... is the phone behind NAT ram ___ --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] SER/OpenSER as registrar to Asterisk (1500 SIP users)
Ser or openser with asterisk can be possible ram On 11/1/07, Antoine Megalla [EMAIL PROTECTED] wrote: Hi, I have a client who requires an Asterisk system with 1500 SIP clients. All clients will have ATAs (mostly Grandstream), so I think a single Asterisk server will not be able to handle all 1500 registrations, plus typical applications like Voicemail, call forwarding, etc.. and the billing needs for all the clients. I have searched all over, and it seems that the perfect solution is using SER/OpenSER as registration server for the SIP clients, and then use Asterisk (one or more servers in load balancing mode) for everything else. The problem is that I cannot find any configuration files for such a setup. I can do all the Asterisk configuration, dial plan, AGIs, apps, etc.. but for SER/OpenSER I cannot find anything. Can anyone please point me in the right direction, provide me with OpenSER configuration, or any pointers on the subject. I tried to read all the material on how to write configuration files for OpenSER, but it is incomprehensible to me, and it is much harder that when I learning Asterisk 3 years ago. Your help is much appreciated. Regards, Antoine Megalla. __ 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] OpenSER for Asterisk Load balance
On 11/2/07, Edgar Guadamuz [EMAIL PROTECTED] wrote:
Hi, guys
I´ve just seen thta OpenSER can be coupled with Asterisk for load
balance, with the dispatcher module, something like this:
dispatcher.cfg file
# group sip addresses of your * units
1 sip:10.1.2.3:5060
1 sip:10.1.2.4:5060
1 sip:10.1.2.5:5060
the basic openser.cfg should be like:
loadmodule(dispatcher.so)
...
if ( method==INVITE ) {
# dst_select( GROUP, HASH METHOD)
dst_select(1,4);
sl_send_reply(100,Trying);
forward(uri:host, uri:port);
exit();
}
That´s OK, but what about failover, I mean, if a Asterisk box crashes,
the dispatcher module will continue sending requests for that IP and
in this case something like heartbeat had to be implemented to take
the failed IP, but it would be more efficient if we could have OpenSER
monitoring the Asterisk servers health, anyone knows how?
Hi
the recomendations are DNS SRV records or
do in round robin basic.
Like failover routes from OpenSER
http://gearsofvoip.com/index.htm
ram
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Re: [asterisk-users] SER/OpenSER as registrar to Asterisk (1500 SIP users)
What about: 1) Message waiting notifications? Especially in a distributed system with multiple Asterisk servers? Openser with Asterisk real time integration can do this job for you 2) Different codecs for different SIP users/accounts? DTMF modes? I know SER doesn't deal with the media at all, but if you let SER handle registrations and authentication, then I'd rather not keep track of codecs/DTMF on asterisk as well. yes OpenSER does not handle media, but you can use dispatcher module to use asterisk as a media relay or Media proxy or rtpproxy can do the job ram ___ --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] SER / Asterisk and mediapath
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can anyone give me any hints? I wasn't sure if using canreinvite since I wasn't sure if that would affect the caller's interaction in the IVR. Hi yes can canreinvite does the job depends on peer compatability ram ___ --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] Meet Me sound file
On 10/29/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was trying to change some of the sound files for the meet me conference application, the one where the user is waiting in the conference with the users waiting in to join (the M option-- enable music on hold when the conference has a single caller) Also what is the name of this sound file? How do I go about changing the file with some other sound file ? Hi look at meetme.conf and you can change the file name and path should be wav file i belive ram ___ --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 Mysql error
On 10/27/07, wassim darwish [EMAIL PROTECTED] wrote: Hi: Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table . I tried to run this command realtime mysql status on the asterisk console and that what i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. Connected to [EMAIL PROTECTED], port 3306 with username root for 9 hours, 43 minutes, 39 seconds. Can any body help with this; Hi what is the version of asterisk and mysql what distro you are using ram ___ --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] Everyone is busy/congested: IP Trunk
On 10/27/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi Pablo; How the IP address will be wrong, and asterisk able to do registeration on the destination? If the IP address wrong, so I will not be able to register on that IP address. Hi i see 2 causes 1. it could be Dialplan issue ( check how the provider accept the call, 1 or just USA number) 2 provider blocked account check network trace to get more info ngrep should be the ideal tool to check the errors in network trace ram ___ --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] Users location --help required
On 10/29/07, srinivas Antarvedi [EMAIL PROTECTED] wrote: Hello all, i am Presently working on integration of asterisk and openser i have a doubt regarding the asterisk . if you take openser when users register it stores the users in location table whether the users running behind NAT or on global ips and when comes to asterisk where does it store ? because i have seen the documentation of integration of asterisk and openser realtime and content there talked about realtime integration of subscriber and sip.conf tables . and i dont want to register users under asterisk so it should fetch the location of users from location table of openser can above fetching mechanism from openser to asterisk using database views be possible? Hi yes Location table is users registred so you can view them in any database ram ___ --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 on Asterisk 1.2.24
On 10/26/07, Steve Totaro [EMAIL PROTECTED] wrote: Does realtime work reliably on Asterisk 1.2.24? Are there any definitive guides, I can only find bits and pieces here and there. Any accurate howtos would be of great help. I am missing func_realtime.so. Where does this file come from? Asterisk or asterisk-addons? I saw in one of the howtos that it is needed. Is it needed for 1.2.X or 1.4.X. Also, what about the switch lines in the .conf files. Some howtos say you need them, others say to delete the whole file, that if for example, extensions.conf exisist, then Realtime wont load extensions. Hi I have not see any problem as of now any of 1.2.X real time, may be you can look integrated package http://voiceone.it/ ram ___ --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 Recording Interface (ARI) integration with Asterisk 1.4
On 10/26/07, Kashif Naeem [EMAIL PROTECTED] wrote: Hello All Has anyone integrated ARI with Asterisk 1.4 ? Is there any manual or steps available ? Also let me know if someone know about any other similar software. Hi Look at Trixbox its already integrated along with asterisk, and ISO image available also ram ___ --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 integration with IBM Sametime
On 10/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I wanted to know if anyone has experience in integration asterisk with IBM Sametime server (by implementing TCSPI). Any pointers for this would be very helpful. Have been reading/googling around a bit and I get to understand that the communication between the Sametime server and Asterisk is SIP. Wanted to know if my understanding is right. Since this is part of some experiment I'm doing, I only have the trial version of Sametime Server with me which doesn't have the Sametime Gateway component (and that is what talks SIP). Just wanted to know if this means that I cannot integrate asterisk with the trial version of Sametime server. Would really help a lot, if someone clarifies my doubts. Hi what are you trying to achieve. Integrating with Asterisk, Samtime server send the calls to asterisk ? or asterisk expect to send calls to Samtime Server ram ___ --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 integration with IBM Sametime
On 10/24/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am trying to setup a conference between Sametime users using conferencing infrastructure of asterisk. Sametime server has a component called TCSPI, which we can implement to talk to any PBX, including asterisk (as per documentation). I was trying to implement the TCSPI for Asterisk. Hi you can configure asterisk to trust any call from Samtime Server and you can configure conference bridge in Asterisk I never tried , but its possible. since iam using 3rd party SIP server, and iam using Asterisk as bridge ram ___ --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] opensr Vs freeswitch SIP proxy server
On 10/23/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have plan for 5000 user register on sip server and call to each other according his/her domain ( Relam ) so which one is best for this type of aaplication or stablity to handle thousand of sip reqest i have study of both product but i need input from community end suggest me best one which can easy and stable for my production my reqierment is [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] [EMAIL PROTECTED] this all domain on my sip server and place all according his domain not interdomain Regards Hi for this kind of things OpenSER is the best, even Freeswitch can do the Job, but OpenSER there since long and testing Million users ram ___ --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] Force codec order
On 10/23/07, Il Neofita [EMAIL PROTECTED] wrote: There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. Hi yes you can do, at client side and as well as Asterisk side. disallow=all allow=first codec allow=second one so on ram ___ --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] opensr Vs freeswitch SIP proxy server
On 10/23/07, satish patel [EMAIL PROTECTED] wrote: dear ram i have also find many document about freeswitch and openser and i thing openser is best then freeswitch it is also module base as well as handle thousand of sip call and easy to impliment with DB but freeswitch is XML base and i am not familer with XML language thats why from my point of view is it taff task Hi i recomend to spend some time and read the documents, and see what is the best to suite your need and find out your own capabilities to deploy the solution. if you feel the task can not achive by you. then opt some cosultant or use some commercial software available to do the best. ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which pci has the dell / hp
On 10/9/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I'm trying to find the right Digium card for the Dell 2950 Dell 2850 HP DL380 G3 HP DL360 G3 Are these 3.3v or 5.0v machines ? I am out of the office, and need to buy a card today. I am looking at either the TE407 or TE412, and would appreciate any help. :) i have tested 3.3v but PCI V 3 if PCI2.X will not work. ram ___ --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] online active call watching
On 9/10/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to Hi with the CDR+mysql you can make query Invite+ack ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sysmaster and Asterisk
On 9/6/07, Mani Nair [EMAIL PROTECTED] wrote: Hello Guys, I am unable to make calls to outside number from some of my extensions. Internally I am able to make and receive calls between extensions and also I am able to receive call from outside number. Any suggestions? Then in am thinking of getting rid of Sysmaster and configure Trixbox to do the entire job that currently my Sysmaster is doing. Any suggestions..? Suggestion is check the dialplan check asterisk cli check network trace with ngrep you have sysmaster and want to move to Trixbox ? ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Debugging to separate log file
On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Dear Jared; I would like to ask if there is a method to let the output of set sip debug ip to be sent for a file? hi when iam doing this i see the server is load is very high how can i send this traffic or mirror traffic to other server and grep the reports ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference in show channels
On 9/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote: 'show channels' shows only running calls while 'sip show channels' shows all running sip sessions including phones trying to register . thanks but after my 30 channels of show channels i see lot of vice break and choppy voice doing passthrough codecs Xeon 2.0GHZ with 2 GG Ram centos 4.4 1.2.17 any suggestions ram On 09/09/2007, ram [EMAIL PROTECTED] wrote: Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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
[asterisk-users] Difference in show channels
Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 or 1.4 for conference call service
On 9/1/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I want to have conference call service and I have A102d sangoma's card so I install asterisk 1.2.x or 1.4.x? Best regards. try DISA, calling card kind ram ___ --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] Distributed System
On 8/29/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Seysan wrote: Is there anywhere that we can look into for Realtime + MySQL that you mentioned? Maybe http://www.voip-info.org/wiki/view/Asterisk+RealTime http://www.asteriskguru.com/tutorials/realtime_pgsql.html Hi any success stories of the setup kindly post your config and information ram ___ --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] app-conference
On 8/28/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I think app-conference is used where there isn't zaptel hardware,in the other word when we use zaptel hardware we shouldn't use app-conference for conference call sevice and we should use meetme application and load ztdummy.Is it true? Best regards. Hi yes app_conference need some timer source app_meetme can use ztummy but on highload expect to use hardware source ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference problem
On 8/23/07, Mark Quitoriano [EMAIL PROTECTED] wrote: Hi, im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running meetme conference, when i try to call meetme i get this from the asterisk console Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No application 'MeetMe' for extension (sample, 65000, 1) i recompiled my zaptel and asterisk, but the app_meetme file still didn't install, what am i missing here? check meetme.conf ram ___ --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] asterisks addon make problem
Hi on debian iam try to make i get this problem any suggestions. make res_config_mysql.so cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o res_config_mysql.o res_config_mysql.c res_config_mysql.c:75: warning: data definition has no type or storage class res_config_mysql.c:77: warning: data definition has no type or storage class res_config_mysql.c: In function âconfig_mysqlâ: res_config_mysql.c:430: error: too few arguments to function âast_config_internal_loadâ res_config_mysql.c: At top level: res_config_mysql.c:463: warning: initialization from incompatible pointer type res_config_mysql.c: In function âunload_moduleâ: res_config_mysql.c:503: error: âSTANDARD_HANGUP_LOCALUSERSâ undeclared (first use in this function) res_config_mysql.c:503: error: (Each undeclared identifier is reported only once res_config_mysql.c:503: error: for each function it appears in.) res_config_mysql.c: In function âparse_configâ: res_config_mysql.c:541: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:548: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:555: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:562: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:569: warning: assignment discards qualifiers from pointer target type res_config_mysql.c:576: warning: assignment discards qualifiers from pointer target type make: *** [res_config_mysql.o] Error 1 ___ --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] asterisks addon make problem
On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 23:08, Tue 21 Aug 07, ram wrote: Hi on debian iam try to make i get this problem What version of Debian? What version of asterisk-addons? Is this an upgrade? We need more info Hi no its fresh installation. asterisk-addons-1.2.7 asterisk-addons-1.2-current.tar.gz Debian 4.0 uname -a Linux 2.6.18-5-686 #1 SMP Sun Aug 12 21:57:02 UTC 2007 i686 GNU/Linux ram ___ --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] asterisks addon make problem
On 8/22/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 00:18, Wed 22 Aug 07, ram wrote: On 8/21/07, Michiel van Baak [EMAIL PROTECTED] wrote: On 23:08, Tue 21 Aug 07, ram wrote: Hi on debian iam try to make i get this problem What version of Debian? What version of asterisk-addons? Is this an upgrade? We need more info Hi no its fresh installation. asterisk-addons-1.2.7 asterisk-addons-1.2-current.tar.gz Debian 4.0 uname -a Linux 2.6.18-5-686 #1 SMP Sun Aug 12 21:57:02 UTC 2007 i686 GNU/Linux Did you install libmysqlclient15-dev ? if not, please do so. -- Hi i have installed that before iam making the addons ram ___ --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] gsm errors
Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions ram ___ --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] gsm errors
On 8/18/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Sat, 18 Aug 2007, ram wrote: Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions This might sound obvious, but Use rfc2833. But apart from that, you're transcoding from GSM (a lossy format) to g729 (another lossy format) so audio quality is going to be quite poor. Can't you stick to either GSM or g729 all the way? What clients are you using that only support GSM? (and if they're on a LAN why not use G711/uLaw/aLaw?) My provider support only g729 and my client have callcenter Suite which support only GSM any suggestion to come over this problem ram ___ --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] AGI SAY TIME
On 8/2/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello all, Can anyone help me with SAY TIME. Every time I ask to say time, it gives me wrong time. I want the system to say time, what ever I give to say. Is it possible? Try to Sync with NTP so the time will not change ram ___ --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 Conference Call
On 7/26/07, satish patel [EMAIL PROTECTED] wrote: Dear all I have asterisk with SNOM SIP phone i want to confrance to my users how to configure confranceing in asterisk meetme.conf is fine but is there any otherway to confranceing If the End device support conference still you can do that ram ___ --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] extension.conf doesn't reload?
On 7/23/07, Devraj Mukherjee [EMAIL PROTECTED] wrote: Hi everyone, I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the reload command in the asterisk command prompt, it doesn't seem to read my configuration files. Any suggestions? pbx*CLI reload The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. == Parsing '/etc/asterisk/cdr.conf': Found [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 == Parsing '/etc/asterisk/http.conf': Found -- Hi when you issue reload command whole asterisk configs are reloaded ( in 1.2.X) but when you reload it says Please use 'module reload' instead may be you try to reload required module ( not tried in 1.4.x) To cross check issue command show dialplan and check your modified config effected or not ram ___ --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] pattern base call routing
On 7/21/07, satish patel [EMAIL PROTECTED] wrote: Than you Hey I have 100 SIP phone with 2 E1 card and IVR feature but i am not happy with my configuration so have u any configuration for advance level Rgd what kind of advanced level Asterisk side or IP phone side ram ___ --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] pattern base call routing
On 7/21/07, satish patel [EMAIL PROTECTED] wrote: i want asterisk extention.conf IVR plan so i want idea of IVR means how other users use IVR in dialplan on asterisk Hi Hint is Look at Agi Scripts you can write small agi scripts to do your job ram ___ --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 load phone registration information
On 7/12/07, Ricardo Carvalho [EMAIL PROTECTED] wrote: Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk thinks those phones are already registered? This would be very usefull for a redundant server... Look at realtime sip should help you ram ___ --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] Codec Negotiation
On 7/12/07, O. Kamal [EMAIL PROTECTED] wrote: I am having a problem with my asterisk gateway, it is accepting only G729, the client is offering G729 and G723.1, however for some reasons, around 15% of calls are rejected due to failed codec negotiation giving an codec error No compatible codecs, not accepting this offer. Anyone gone through this before? you can allow UA to accept other codecs by adding allow=ulaw.others in sip.conf ram ___ --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 , upgrade asterisk
Yeah, the only time you should delete *everything* from the modules directory, is when upgrading between major versions, such as from 1.2 to 1.4. what happend this situation still i need to re-register or just copy the g729 of 1.4 and copy the license will this work? ram ___ --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] Avoided deadlock for '0x864e70', 10 retries!
Hi iam using 1.2.X SVN iam keep getting the below message Jun 28 23:07:31 WARNING[2692]: channel.c:785 channel_find_locked: Avoided deadlock for '0x864e70', 10 retries! any help ram ___ --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] g729 problem
Hi iam using asterisk 1.2 version I have purchased g729 license from Digium when iam making calls, iam getting this error ? Jun 25 14:41:45 NOTICE[4424]: frame.c:183 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end any help ram ___ --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 problem
On 6/25/07, Karl J. Vesterling [EMAIL PROTECTED] wrote: Disable Voice Activity Detection yes i have disabled at my eyebeam, still i see this error iam using 1.2.18 ram ___ --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 0 dial outgoing call
On 6/22/07, satish patel [EMAIL PROTECTED] wrote: Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-[*][mediant2k]-[Avaya_PBX]--e1-[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give dialtone from pstn how to setup extention.conf for outside call create dialplan for the same ram -- Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --Bandwidth and Colocation provided by Easynews.com http://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] Query
On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of modutils and iptable. Can anybody help me out of this. Thanx and Regards sanchal singh Either contact digium support or post the problem ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] config files to mysql convertion
Hi does any one come acrosss the tool which convert the normal config files of sip.conf, extension.conf...etc will convert automatically to mysql. with with any problems if yes, kindly point me to that toolm which iam looking thanks ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme realtime
On 6/7/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Thu, 2007-06-07 at 11:02 +0530, ram wrote: is this possible ? You can only do it with realtime static. how can i do that, any document URL to achieve that ram Hi I have read that, but i dont see any examples that give me solution for meetme. can you just give me some examples ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme realtime
On 6/7/07, Carlos Chavez [EMAIL PROTECTED] wrote: On Thu, 2007-06-07 at 21:41 +0530, ram wrote: I have read that, but i dont see any examples that give me solution for meetme. can you just give me some examples I think the example shown on that page, even though it is for extensions.conf, is very clear. Just put the context into category, what comes before the = into var_name and what comes after into var_val. extconfig.conf: meetme.conf = mysql,asteriskcdrdb,ast_config to insert a meetme room do: INSERT INTO ast_config SET filename='meetme.conf', category='rooms', var_name='conf', var_val='900'; Basically var_val stores: confno[,pin][,adminpin] let me try a chance and get back to you if any problem thanks ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme realtime
Hi iam using 1.2.17 does any one have information meetme in realtime and store in mysql i dont see any document could some one help me is this possible ? ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme realtime
is this possible ? You can only do it with realtime static. how can i do that, any document URL to achieve that ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] debug logs
Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:49 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:49 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:50 DEBUG[2173] manager.c: Manager received command 'Command' ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 License
On 6/4/07, Arun Kumar [EMAIL PROTECTED] wrote: HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? Hi no its bound to ethernet address, when you change ethernet you need to register again with support of digium its only use for one Server ram thanks arun ___ --Bandwidth and Colocation provided by Easynews.com http://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] debug logs
This notifies you that it has been used (IIRC). Hi what does that mean , it has been IIRC ? ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 client and server Side
Hi iam using G729 at server side and same iam using eyebeam with g729 at client side still its take transcoding CPU process or its pass through ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme problems
Hi I have reading the voiip side i found some document says The conference bridge runs Ulaw codec by default. If you let people connect with GSM or other codecs, Asterisk will use CPU power to convert audio between codecs iam using vicidial and meetme for callcenter application. iam geting choppy voice, and voice breaks. iam using connecting VoIP SIP provider using g729 codec, since i can save bandwidth iam using client side also g729, so no translation required but after i see this document, will meetme convert the g729 to GSM or ULAW internall, and i have will have cpu load, is this correct. if i dont want to CPU loadup more, i should use GSM or ULAW at client side is this correct. can some one correct me if iam wrong suggestions welcome ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38
Hi iam using asterisk1.2.18 in the logs i keep getting this message any help ram Jun 2 05:47:41 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38 Jun 2 05:47:41 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38 Jun 2 05:54:00 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39218 udptl t38 Jun 2 05:54:00 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39218 udptl t38 Jun 2 05:54:45 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39678 udptl t38 Jun 2 05:54:45 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39678 udptl t38 Jun 2 05:54:54 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39706 udptl t38 Jun 2 05:54:54 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 39706 udptl t38 Jun 2 06:01:09 WARNING[890] app_meetme.c: Unable to write frame to channel SIP/1006-b7803008 Jun 2 06:05:21 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:21 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:22 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:22 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:22 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 Jun 2 06:05:23 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 35188 udptl t38 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel_find_locked: Avoided deadlock
Hi iam still getting this error does any one suggest me what is need to be done WARNING[2898]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8327c0', 10 retries! after this my voice quality going very low voice is going choppy any suggestions ram On 5/30/07, Rob Schall [EMAIL PROTECTED] wrote: I too have this problem. I have two queues set up, and one is in use. I didn't realize thats what caused those errors. I am also using sip. Here are my setups if it helps anyone find a bug: Queues.conf [billing] music=default strategy=ringall reportholdtime = no timeout=8 retry=10 wrapuptime=10 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = Agent/3876 member = Agent/5055 member = Agent/8318 member = Agent/8323 member = Agent/8324 Agents.conf ;Billing agent = 3876,,Christina agent = 8318,,Stephanie agent = 8323,,Rob agent = 8324,,Colleen agent = 5055,,Chris Extensions.conf exten = s,1,Answer() exten = s,n,Ringing() exten = s,n,Wait(2) exten = s,n,Queue(billing,t|||30) exten = s,n,Voicemail(u) exten = s,n,Hangup() ram wrote: On 5/30/07, Jaswinder Singh [EMAIL PROTECTED] wrote: Is it over iax and there are lot of outgoing channels ? If yes then you are not the only person having this .. SIP ram -- ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://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] channel_find_locked: Avoided deadlock
Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channel_find_locked: Avoided deadlock
On 5/30/07, Jaswinder Singh [EMAIL PROTECTED] wrote: Is it over iax and there are lot of outgoing channels ? If yes then you are not the only person having this .. SIP ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAPTEL problem
Hi I have 100XP Digium clone card Installed in my pc and compiled zaptel and asterisk again after installing the card but after i rebooted i can load zaptel and wcfxo modprobe with out any problem but when i intiated ztcfg - i get the following error Zaptel Configuration == 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) - dmesg errors Zapata Telephony Interface Unloaded Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r1725 Echo Canceller: KB1 Failed to initailize DAA, giving up... wcfxo: probe of :00:10.0 failed with error -5 -- lspci 00:0c.1 SCSI storage controller: Adaptec AIC-7896U2/7897U2 00:10.0 Communication controller: Motorola Unknown device 5608 any suggestions ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Time Card
On 5/26/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Shanon and everyones input... Finally, got the application working as planned with PHPAGI... Now the only draw back is the voice... I am using text2wav to prompt all the questions, but the voice is creepy... Is their any easier way to replace the text2wav voice with proper recorded female voice? Please advice... what codec are you using ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Time Card
On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, I have been looking for this solution for quite sometimes Asterisk Time Card System. I found some discussion from Digium forum but not quite helpful. Hi what is the mean of time card system ? is this kind of attendent system ? kindly give some more details ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Clusters
On 5/23/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, I need to implement a clustered PBX System where parent * is connected to one of the outbound carrier and other child * will register to parent *. Reason for this implementation is because some of the child * are behind NAT. Parent * is on Public IP Address and its connected to outbound carrier. Child * will only send out long distances calls to Parent * to terminate, rest are internal calls. Now which is the best way to implement this type of scenario... DUNDi? or custom context? Hi why dont you looking this kind of solution DS3TDMOVERIPSER-ASterisk ram Thanks, Nitesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Time Card
On 5/24/07, David Gomillion [EMAIL PROTECTED] wrote: On 5/24/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks for your reply, The basic system would work as follows: - Method 1 === An employee would call in to the system and a welcome message is prompted. After that a employee is asked to enter the employee ID and PIN number and once verified Employee ID, Caller ID, and time of day is stored into MySQL DB. By end of the day employee will call in again to logout from the system and same information is stored into the DB. Method 2 === This time employee is verified with Caller ID, so the employee ID and PIN number is skipped and time of day is logged into the DB. Is it possible? Thanks, Nitesh Anything is possible. But I haven't seen one off-the-shelf. It really won't be a big deal to write, though. We created a timeclock application and toyed with allowing people to clock in via phone, and I even wrote the extension logic, but we opted to not enable it because we don't trust our employees that much. This was years ago, when we were running pre-1.0 code. We've switched servers a few times, so the logic is long gone, but it only took an afternoon to write and debug. with the AGI you can do all ram ___ --Bandwidth and Colocation provided by Easynews.com http://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] Working softphone for poket PC
On 5/23/07, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Windows Mobile 6 comes with a SIP client, however on my HTC device I still need to use the speaker phone or a headset, the GSM phone speaker won't do: http://thinkabdul.com/2007/04/25/sip-config-loader-free-utility-to- automatically-configure-load-_setupxml-file-for-sip-voip-on-windows- mobile-60-device/ Other clients that I haven't tested yet (apart from SJphone - how do you register, I only manged to do URL dialing?): * Express Talk (free, http://www.nch.com.au/talk/ptalksetup.exe) * Kapanga (beta?) * voipsurfer (IAX, not free) * ppciax (IAX) * eScSoftphone (IAX, Demo available, http://www.electronicscience.com/) * agephone * gphone * x-pda * iFon (SIP, H.323, Video, Messaging, www.voip-info.org/wiki/view/iFon) HI any softphone for my sony erricson p990i SE says that its got SIP support but i dont see their releases or does ny one have source codes, for UIQ3 ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to connect two asterisk server
On 12/31/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Carlos, Im interested in knowing how we can connect 2 server using SIP. Well for me both are not asterisk servers, 1 is asterisk and 2nd is an SIP based Server. i need to take multiple calls from the SIP based server and terminate it using my asterisk peers based on my dialplan. I can use SIP only as my other server doesnot support IAX2. How can i get that. Please let me know Hi what does it mean, sip based server ? please do mention what is that server, most of the servers are SIP based only. ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
