[asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation
Hello everyone, As the most established implementer of Asterisk in Singapore, we often have a need to provision hundreds of IP phones at a time. Provisioning each IP phone by editing each xml file individually was very time consuming so we developed a tool internally called IP Phone Provisioning Tool with which we now deploy hundreds of phones in a matter of a few minutes as opposed to a few hours or days. We decided to release this tool to the Asterisk community as a freeware. It works with all IP phones which use xml files. It runs on any modern flavor of Windows. If you are interested in it you can download a copy at http://www.voip.com.sg/voip_ip_phone_provisioning_tool.html . Any suggestions can be directed to me at [EMAIL PROTECTED] . If you like the tool, do spread the word by mentioning us in your website or blogs. With Regards, San Lantone Information Systems LLP Tel : SG +65 62271149 (Ext 958) US +1 646 8621550 (ext 958) UK +44 207 0239247 (ext 958) Visit our websites to learn more about our products : www.voip.com.sg (Learn more about VOIP and how we can help you implement it) www.mailtracking.com (Track your emails, featured in Channel News Asia and various other publications around the world) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk MSOutlook Dialer
Hello everyone, we just wrote a little MSOutlook address book dialer interfaced with Asterisk. It is a small (400k) exe that you need to install. It is completely free to use, either for educational purpose or otherwise. You can download it at http://www.voip.com.sg/voip_products/voip_asterisk_outlook_dialer.html . Please send your comments to me directly as I am the developer for it. With Regards, Sandeep Singhania Lantone Information Systems LLP Tel : SG +65 62271149 (Ext 958) US +1 646 8621550 (ext 958) UK +44 207 0239247 (ext 958) Fax : +65 68750242 Mobile: +65 97471958 Visit our websites to learn more about our products : www.voip.com.sg (Learn more about VOIP and how we can help you implement it) www.mailtracking.com (Track your emails, featured in Channel News Asia and various other publications around the world) www.callaccounting.ws (Home of the world's most popular Call Accounting System proudly developed by us) ___ --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] Toll free nos
Hello everyone, I am in need of 20 US toll free nos and 10 non toll free nos, termination using IAX. Are there any reliable companies that you can recommend? Thank you With regards, San ___ --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] Digium TE205 Card
Hello Everyone, I have a brand new Digium TE205 card, bought 2 days back, stillunopenedand for sale. Reason for selling is we need a quad span ISDN card now instead of dual span ISDN card. Selling it at USD700, this card retails at around USD900+ right now.Card was directlypurchased from Digium and their invoice will be supplied as proof. If you are interested, leave me an email at [EMAIL PROTECTED]or call me at 718 2336260x 120. With regards, San ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk based Call Accounting software - 1st release
Hello Asterisk community, After numerous request from various companies where we have implemented * as a phone system and also from many other * users all over the world,yesterday wereleased the 1st version of Asterisk module for Call Accounting Mate (www.callaccounting.ws) . As some of you know we also use Asterisk internally as our phone system and as developers for Call Accounting Mate, we felt it was necessary to implement a decent Call Accounting software for *. Call Accounting Mate runs on Windows and is completely web based. It ships with the necessary source files andAsterisk modules to interfaceAsterisk via tcpip to Call Accounting Mate. We have set up a Asterisk - Call Accounting Mate forum so we can gather input from the Asterisk community. You can access the forum at http://www.callaccounting.ws/forum/index.php?board=5.0. Regards, San Singhania www.callaccounting.ws Tel : +1 718 5762066 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting DTMF reliably
Hello everyone, I am having big problems trying to detect dtmf tones while a IVR prompt is playing on zap channels. Sometimes the detection only starts 4-5 seconds into the prompts. Other times it works very well for the 1st few calls and then starts having problems. And most times it also does not detect all digits, eg when 123 is keyed in, it may detect 23 or 2 or 3, but never the complete string. Can someone help me with this? I am in Singapore so I dont know if its a localisation issue. Also I have relexdtmf on and my rxgain and txgain on zap channels are set to 0 so i dont think thats the problem. Help is most appreciated as I cannot go on with Asterisk if this function does not work :( Thanks San ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting DTMF tones
Hello everyone, I am having big problems trying to detect dtmf tones while a IVR prompt is playing on zap channels. Sometimes the detection only starts 4-5 seconds into the prompts. Other times it works very well for the 1st few calls and then starts having problems. And most times it also does not detect all digits, eg when 123 is keyed in, it may detect 23 or 2 or 3, but never the complete string. Can someone help me with this? I am in Singapore so I dont know if its a localisation issue. Also I have relexdtmf on and my rxgain and txgain on zap channels are set to 0 so i dont think thats the problem. Help is most appreciated as I cannot go on with Asterisk if this function does not work :( Thanks San ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iconnect and Asterisk
Hello All, I have gone thru all the resources I could find on google on asterisk + iconnect and managed to get outgoing calls working. However, I cannot get incoming calls to work at all. With the sip debug on, I can see that something is happening everytime a call is received from iconnecthere, but I get an invalid tone on the caller side. The call never rings anywhere on the asterisk. Would appreciate any help on this. Thanks Below is my sip file register=442087926805:[EMAIL PROTECTED]:5060 [iconnecthere]type=friendsecret=somepasswordusername=11232634host=sipauth.deltathree.comcanreinvite=no;nat=yescontext=default;dtmfmode=inbanddisallow=all;allow=allallow=gsmallow=ulawallow=alawallow=g726allow=g723 This is the sip debug info when a call comes in from iconnecthere : 11 headers, 0 linesReliably Transmitting:REGISTER sip:sipauth.deltathree.com SIP/2.0Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9From: sip:[EMAIL PROTECTED];tag=as5c70755cTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 104 REGISTERUser-Agent: Asterisk PBXExpires: 120Contact: sip:[EMAIL PROTECTED]Event: registrationContent-Length: 0 (no NAT) to 213.137.73.140:5060localhost*CLI Sip read:SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9To: sip:[EMAIL PROTECTED]From: sip:[EMAIL PROTECTED];tag=as5c70755cCall-ID: [EMAIL PROTECTED]CSeq: 104 REGISTERContent-Length: 0 7 headers, 0 lineslocalhost*CLI Sip read:SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK35628ab9From: sip:[EMAIL PROTECTED];tag=as5c70755cTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 104 REGISTERContact: sip:[EMAIL PROTECTED]:5060;expires=120Contact: sip:[EMAIL PROTECTED]:5060;expires=14Expires: 120Content-Length: 0 10 headers, 0 linesDestroying call '[EMAIL PROTECTED]'localhost*CLI Sip read:INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4-1Via: SIP/2.0/UDP 213.137.81.27:5060;received=213.137.81.27To: sip:[EMAIL PROTECTED]From: sip:[EMAIL PROTECTED];tag=DF81964C-1341Call-ID: [EMAIL PROTECTED]CSeq: 101 INVITEContact: sip:[EMAIL PROTECTED]:5060Record-Route: sip:[EMAIL PROTECTED]:5060;maddr=213.137.73.173Record-Route: sip:[EMAIL PROTECTED].27:5060;maddr=213.137.73.176Content-Type: application/sdpContent-Length: 146 v=0o=CiscoSystemsSIP-GW-UserAgent 5851 2446 IN IP4 213.137.81.27s=SIP Callc=IN IP4 213.137.81.27t=0 0m=audio 18958 RTP/AVP 4 0 8 2 101 13 headers, 6 linesUsing latest request as basis requestSending to 213.137.73.140 : 5060 (non-NAT)Found RTP audio format 4Found RTP audio format 0Found RTP audio format 8Found RTP audio format 2Found RTP audio format 101Peer audio RTP is at port 213.137.81.27:18958Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x1d(G723|ULAW|ALAW|G726)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)Found peer 'iconnecthere'Reliably Transmitting (no NAT):SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4-1Via: SIP/2.0/UDP 213.137.81.27:5060;received=213.137.81.27From: sip:[EMAIL PROTECTED];tag=DF81964C-1341To: sip:[EMAIL PROTECTED];tag=as34968f1dCall-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Proxy-Authenticate: Digest realm="asterisk", nonce="252c7e0a"Content-Length: 0 to 213.137.73.140:5060Scheduling destruction of call '[EMAIL PROTECTED]7' in 15000 mslocalhost*CLI Sip read:ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.173Via: SIP/2.0/UDP 213.137.73.176:5060;branch=34550e33-69d4c647-76eb3474-c49105c4-1From: sip:[EMAIL PROTECTED];tag=DF81964C-1341To: sip:[EMAIL PROTECTED];tag=as34968f1dCall-ID: [EMAIL PROTECTED]CSeq: 101 ACKContent-Length: 0 8 headers, 0 lineslocalhost*CLI Sip read:REGISTER sip:192.168.1.250 SIP/2.0Via: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42caFrom: sip:[EMAIL PROTECTED];tag=d96a1d9a3a8eb4afTo: sip:[EMAIL PROTECTED]Contact: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 402 REGISTERExpires: 120User-Agent: Grandstream BT100 1.0.4.67Max-Forwards: 70Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBEContent-Length: 0 12 headers, 0 linesUsing latest request as basis requestSending to 192.168.1.60 : 5060 (non-NAT)Transmitting (no NAT):SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.1.60;branch=z9hG4bKda87dee87d2b42caFrom: sip:[EMAIL PROTECTED];tag=d96a1d9a3a8eb4afTo: sip:[EMAIL PROTECTED];tag=as5926604eCall-ID: [EMAIL PROTECTED]CSeq: 402 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]Content-Length: 0 to
[Asterisk-Users] Rebooting Linux / Asterisk
Hello everyone, I am new to Linux, some help with the following would really beappreciated : 1.How can I load asterisk automatically in Linux each time the machineboots up (like autoexec.bat in windows)2.how I can shut down and restart asterisk automatically every night? Thanks San ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Call Manager and Asterisk (AVVID) - Comparison
Hello All, I have a large customer with close to 30 offices worldwide who want a VOIP solution. We have already implemented it in Singapore and India, but as usual the Corporate IT department which is inLondon is pushing for the Cisco call manager for all their locations. I dont know anything much about it. Is there a paper or comparision on this? I am sure many asterisk implementors must have gone down this road before so it would be nice if they can point me to the right direction. Thanks San
[Asterisk-Users] Dial Fail - Send Email
Hello, I have an asterisk implementation that is running for the last 2 months. Now the customer wants to be able to get an email everytime a dial command fails...i.e when either no one picks up, its busy or the link to the end user device is down. Actually, this is a small call centre type of installation. * is located in Singapore and the end points (i.e agents) are located in India. And the reason he wants this email (not voicemail notification to email, just a notification that the call did not get thru) is because the link between * and the agents may be down.mainly due to the internet connectivity issues. Does anyone have such an app? Also, this app should send multiple email addresses the email that the link may be down. If there isn't such an app, can someone develop it for me? I will be willing to pay a small price for it. Thanks San
[Asterisk-Users] SMDR/CDR - Asterisk integration
Hello everyone, I am developing an online SMDR / call log system for asterisk. This is going to take the form of an executable with embedded sql and webserver, pdf generation, excel generation, graphs. Actually, we have been selling this for a while now with great success and now I am starting work on the integration with Asterisk. Its a windows executbale and the executable is just about 1MB. If someone is interested, let me know. The online demo is at http://demo.callaccounting.ws . The username/password is admin and admin. To print out reports, just leave all the fields for the report selection blank. With regards, San
[Asterisk-Users] Asterisk Article
Hello, sometime back, I saw an article (i think it was in powerpoint) comparing Asterisk for suitability for call centres. This article has features on one side and tick for each feature that * has. I cant seem to find that article. Infact I have no idea where I saw it except it was on the internet. I have searched on Google but come out blank. Need someone's help on it. Can someone point me to the right direction please. Thanks San
[Asterisk-Users] Error compiling Zaptel
Hi, I just finished downloadingasterisk and when trying to compile the zaptel drivers, get the following errors. I dont have a clue whats going on... can someone help. In file included from /usr/include/linux/module.h:20,from zaptel.c:44:/usr/include/linux/modversions.h:1:2:#error Modules should never use kernel-headers system headers,/usr/include/linux/modversions.h:2:2: error but rather headers from an appropriate kernel-source package./usr/include/linux/modversions.h:3:2: #error Change -I/usr/src/linux/include (or similar) to/usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname -r)/build/include/usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel.make: *** [zaptel.o] Error 1 Thanks San
[Asterisk-Users] X100P card issues - noise, volume, etc
Hello, I have just managed to get my 1st * server up and running and have a lot of issues with theX100P analog card. Would really appreciate anyone trying to help me on the following : 1. The receive and transmit is too soft. So i increased the txgain and rxgain. The volume is fine after this, but there is a lot of 'wind' noise on the line. I have my echo cancellation on, aggregive suppression on but still no use. However, the moment I set the txgain and rxgain back to 0, everything works fine but both parties compain that they cannot hear each other. Any solution for this? Also, if I use an ISDN interface, will it solve my problems? I understand that since it is digital, there are no longer issues with hangup detection, noise, echo, etc. Is that true? Also, has anyone had any experience with conferening (i.e bridging) 2 FXO cards together? Thanks San
[Asterisk-Users] Sipura SPA-2000
Hello, I am very new to asterisk and voip in generaland so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I want to know is that if I can use the SPA 2000 as a FXO. And if I can, are there any issues I need to be aware of. I am going to rely very heavily on conferencing so if the SPA 2000 can be used as a FXO, will it be suitable for this purpose? Thanks San
