Re: [asterisk-users] Drag and Drop transfer application
Hello, On Thu, Apr 24, 2008 at 9:24 PM, Al lists [EMAIL PROTECTED] wrote: any of you guys have used FOP for drag and drop transfer on 30 40 phones environment? how stable is that? I'm playing with it but so far drag and dropping phone icon to another phone disconnectes the call. If your calls disconnects it is probably a misconfiguration (the asterisk CLI with some debug and verbose levels will help you find out). You have to match a context and extension as defined in op_buttons.cfg with the [EMAIL PROTECTED] you use in your dialplan. Also you have two ways or points of view to perform transfers, dragging the other leg (default behaviour), or your own.. you have to select which one you like with reverse_transfer=1 in op_server.cfg Best regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ -- Bandwidth and Colocation Provided 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] Upgrading to 1.4
On Sat, 26 Apr 2008, Rob Hillis wrote: As is just about always the case, posting twice to the list within three hours is not only unlikely to get a faster response, I would in fact imagine it would /reduce/ your chances of getting a response at all. I suspect he didn't. I've seen many instances here where posts appear twice (or more). In my own experience of running mailling lists over the years, I've found this is often caused by some broken Exchnge server somewhere lotusscript wrote: A good while back when installing 1.2 there were major issues with UK callerid. Asterisk 1.2 didn't recognise the callerid correctly because of the way BT sent the information. Sometimes before the first ring or just after. After applying a third party patch we got it to work. We were afraid to touch it after that :-) Has this problem now gone away with 1.4? No idea about this though - I had the same issues, but have stuck to 1.2 for the time being. (And from what I recall, it was nothing to do with the way BT sent the information, but the wctdm driver simply being broke until the patch fixed it) Gordon ___ -- Bandwidth and Colocation Provided 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] Upgrading to 1.4
On 25 Apr 2008, at 15:58, lotusscript wrote: A good while back when installing 1.2 there were major issues with UK callerid. Asterisk 1.2 didn't recognise the callerid correctly because of the way BT sent the information. Sometimes before the first ring or just after. After applying a third party patch we got it to work. We were afraid to touch it after that :-) Has this problem now gone away with 1.4? I run Asterisk 1.4.19 with Zaptel 1.4.10 on Ubuntu with a TDM400P - and still having issues with Callerid and Distinctive Ring. The only way I have managed to get callerid to work with success is to patch the Zaptel source in 1.4.5.1 - however distinctive ring is broken - but this appears to be a Chan_Zap issue rather than Zaptel. Regards Matt ___ -- Bandwidth and Colocation Provided 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] Need comments on CRM development / Asterisk Customization
Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? Regards -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided 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] Need comments on CRM development / Asterisk Customization
Vicidial has call center / CRM integration with asterisk ... with many years of bug reporting ... it open source. On Sat, Apr 26, 2008 at 12:11 PM, Kashif Naeem [EMAIL PROTECTED] wrote: Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? Regards -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided 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] Need comments on CRM development / Asterisk Customization
AJAX! On Sat, Apr 26, 2008 at 8:26 AM, Arthur [EMAIL PROTECTED] wrote: Vicidial has call center / CRM integration with asterisk ... with many years of bug reporting ... it open source. On Sat, Apr 26, 2008 at 12:11 PM, Kashif Naeem [EMAIL PROTECTED] wrote: Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? Regards -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
On Fri, Apr 25, 2008 at 4:15 PM, Matt Florell [EMAIL PROTECTED] wrote: On 4/25/08, Jared Smith [EMAIL PROTECTED] wrote: On Fri, 2008-04-25 at 18:48 +, Arthur wrote: I still hope someone would enlighten us by his experience in doing call recordings without recording to RAM Drive. I can't speak for Steve's solution (as I'm not sure exactly what he's doing) but I could take a stab in the dark and guess that he's capturing the audio at the network layer (on a completely different box than Asterisk is running on) and recording it from there. But that's just a guess... To address several points: OrecX (http://www.orecx.com/) can do call recording outside of the Asterisk core using several different methods depending on your needs and channeltypes. In fact even with Sangoma TDM cards you can capture audio at the kernel level and send the audio as RTP streams very efficiently(3% CPU load for 92 channels) to an OrecX server on your network. It must be mentioned that setting up Orecx with retrieval might be a little complex for some Asterisk users, especially if you are recording a large amount of calls, or are recording on more than one Asterisk server, and if you choose this route you would do well to hire an experienced consultant(or contact Oreca directly) to do the install for you. As far as Asterisk-based recording, writing to a RAM drive(or tmpfs) is about your only option if you are planning on doing more than 50 concurrent recordings, if you are using Asterisk it is a viable and tested solution. I have several client systems that are recording well over 50 calls concurrently on a daily basis this way. If you will be recording directly to hard drives with any frequency or volume I would strongly recommend NOT using standard IDE or SATA hard drives, they burn up and fast. Use a caching SCSI drive controller with some high quality SCSI drives and you can record to those drives for years even at 40 concurrent channels recording all day every day. Hope that helps, MATT--- I paid for the OrecX ability to save the recordings as the sipcallid. This is fairly easy to track and match up in a CRM so long as you are writing to a DB. Before that you just had a bunch of folders based on day and hour and the filenames were impossible to track, IP address and time I believe. Not much use when recording 15k calls a day. I also worked with Bruno @ Oreca to get their passive recording solution from it's infancy (~10 or so concurrent calls) to a real enterprise solution (maxing at ~200 concurrent recordings per server). Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to retrieve sip tag from dialplan
Hi to all is it possible to retrieve the sip tag (server side) of a sip call from the dialplan? Thanks. -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
OK, I think you need to home in on the differences between the server(s) that work fine and the one that doesn't. As I said in my other mail, the faulty one is a .. mono processor machine, with SMP turned on .. running CentOS 5 .. with kernel : 2.6.18-53.1.13.el5 There are other kernels too(2.6.18-8, etc.), will be trying those kernels too. The local working machine is : .. dual processor, with SMP ofcourse .. running Fedora Core 7, if I remember it correctly. .. kernel definitely 2.6.13 Have looked at all parameters, be it the kernel timer frequency(1000 HZ), enhanced timer support, etc. Everything seems to be set right. (Then again, I hope I am looking at the correct places, i.e. .config files and using make menuconfig). Try watch -dn 1 cat /proc/interrupts and check that the RTC interrupts are going up by 1024 per second. This is with ztdummy running. This I gotta try. What if it isn't? And worse, what if it is and I am still getting the choppy playbacks!! What else is going on on this server? Does it have any virtual machines on it? Does it have X Windows running? What does top show? Unfortunately a lot of other processes are running too on the server, one of them being httpd and other sundry needed by the client (this inspite of suggesting him to otherwise). This is an Asterisk install not done by me, I just added the zaptel installation and ztdummy module. Was brazenly confident of things working in a jiffy(does this count as a pun?), when I stepped in. cheerz :-( - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided 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] Need comments on CRM development / Asterisk Customization
On Sat, 2008-04-26 at 15:11 +0300, Kashif Naeem wrote: Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? Ever looked at sugarcrm ? For SuSE it is in the add-on repositories: .../repo/home:/vinboy/openSUSE_10.3/src/sugarcrm-5.0.0b-26.1.src.rpm .../repo/home:/vinboy/openSUSE_10.3/noarch/sugarcrm-5.0.0b-26.1.noarch.rpm info: http://www.sugarcrm.com/crm/ (commercial opensource) hw ___ -- Bandwidth and Colocation Provided 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] Roaming callback?
Hi list, Regarding callback functionality, it seems that Asterisk only includes a provision for callback in the voicemail configuration, for authorization purposes, but not an actual callback mechanism. For that, there are various free 3rd party AGI (Asterisk Gateway Interface) scripts available: * Asterisk tips callback http://www.voip-info.org/wiki/view/Asterisk+tips+callback * capi Callback http://www.junghanns.net/en/callback.html Looking at the scripts, they don't seem too difficult to implement, but they don't exactly work as I was hoping either. First, they require your system to have a dedicated callback number that you ring once and then hang up. The system then calls you back at a predefined number, e.g. your mobile phone. Not a very flexible solution. What I had in mind was an option in the voicemail menu that would allow you to dial a number -- any number -- at which the system would call you back. I'd call this roaming callback. Is anything like this available for Asterisk? Cheers, Jaap ___ -- Bandwidth and Colocation Provided 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] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
On Sat, Apr 26, 2008 at 9:14 AM, Benjamin Jacob [EMAIL PROTECTED] wrote: OK, I think you need to home in on the differences between the server(s) that work fine and the one that doesn't. As I said in my other mail, the faulty one is a .. mono processor machine, with SMP turned on .. running CentOS 5 .. with kernel : 2.6.18-53.1.13.el5 There are other kernels too(2.6.18-8, etc.), will be trying those kernels too. The local working machine is : .. dual processor, with SMP ofcourse .. running Fedora Core 7, if I remember it correctly. .. kernel definitely 2.6.13 Have looked at all parameters, be it the kernel timer frequency(1000 HZ), enhanced timer support, etc. Everything seems to be set right. (Then again, I hope I am looking at the correct places, i.e. .config files and using make menuconfig). Try watch -dn 1 cat /proc/interrupts and check that the RTC interrupts are going up by 1024 per second. This is with ztdummy running. This I gotta try. What if it isn't? And worse, what if it is and I am still getting the choppy playbacks!! What else is going on on this server? Does it have any virtual machines on it? Does it have X Windows running? What does top show? Unfortunately a lot of other processes are running too on the server, one of them being httpd and other sundry needed by the client (this inspite of suggesting him to otherwise). This is an Asterisk install not done by me, I just added the zaptel installation and ztdummy module. Was brazenly confident of things working in a jiffy(does this count as a pun?), when I stepped in. cheerz :-( - Ben. http://www.openvox.com.cn/products_detail.php?genre_id=9id=28 If you can get the bare card, you can use it for timing with a little magic that can be found via google. If not, get one with an FXO or FXS and you will add a little flexibility and have real hardware timing. If you continue to have issues, then you can eliminate timing and focus on processes I would think. I had a client running spamassassin on their Asterisk box which doubled as their corporate email server, geewhiz, I wonder why they were having issues. Another odd thing Tzafrir helped me to notice was (I don't remember what version of CentOS) that the time was jumping ahead a couple of minutes and then back. Running top, you could tell something was up because it was refreshing way too fast. Then typing date on the command line repeatedly showed the time jumping all over the place. Might want to check that out too. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outside call not coming through
When i try to call '36946811' from the outside the call gets through, but is rejected and the sound file is not played, this is my conf and sip debug output: ## sip.conf [general] context=incoming register = 36946811:[EMAIL PROTECTED]/1234 port=5060 bindaddr=0.0.0.0 srvlookup=yes ## extensions.conf [incoming] exten = 36946811,1,Background(hello-world) ## sip debug *CLI --- SIP read from 87.54.25.114:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:87.54.25.114;ftag=688c7f1d;lr=on Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0 Via: SIP/2.0/UDP 192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060 Max-Forwards: 16 Contact: sip:[EMAIL PROTECTED]:5060 To: sip:[EMAIL PROTECTED] From: Harrysip:[EMAIL PROTECTED];transport=UDP;tag=688c7f1d Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp User-Agent: Zoiper rev.417 Content-Length: 311 v=0 o=Z 0 0 IN IP4 192.168.2.5 s=Z c=IN IP4 192.168.2.5 t=0 0 m=audio 8000 RTP/AVP 3 110 97 8 0 101 a=fmtp:97 mode=30 a=fmtp:101 0-15 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=direction:active - --- (14 headers 15 lines) --- Sending to 87.54.25.114 : 5060 (no NAT) Using INVITE request as basis request - NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. Found peer 'musimi' --- Reliably Transmitting (NAT) to 87.54.25.114:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0;received=87.54.25.114 Via: SIP/2.0/UDP 192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060 From: Harrysip:[EMAIL PROTECTED];transport=UDP;tag=688c7f1d To: sip:[EMAIL PROTECTED];tag=as0f99b309 Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=35c07307 Content-Length: 0 Scheduling destruction of SIP dialog 'NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.' in 32000 ms (Method: INVITE) --- SIP read from 87.54.25.114:5060 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0 From: Harrysip:[EMAIL PROTECTED];transport=UDP;tag=688c7f1d Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. To: sip:[EMAIL PROTECTED];tag=as0f99b309 CSeq: 2 ACK Content-Length: 0 - --- (7 headers 0 lines) --- Reliably Transmitting (NAT) to 62.107.1.48:5060: OPTIONS sip:[EMAIL PROTECTED]:5060;rinstance=d815b062f3a40a5e SIP/2.0 Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport From: asterisk sip:[EMAIL PROTECTED];tag=as00c1a604 To: sip:[EMAIL PROTECTED]:5060;rinstance=d815b062f3a40a5e Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 26 Apr 2008 14:46:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- --- SIP read from 62.107.1.48:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport=5060 Contact: sip:192.168.2.5:5060 To: sip:[EMAIL PROTECTED]:5060;rinstance=d815b062f3a40a5e;tag=d4846e53 From: asterisksip:[EMAIL PROTECTED];tag=as00c1a604 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS User-Agent: Zoiper rev.417 Allow-Events: message-summary Content-Length: 0 - --- (13 headers 0 lines) --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS ___ -- Bandwidth and Colocation Provided 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] Outside call not coming through
Screwed up really bad. This is the correct config and sip debug: ## sip.conf [general] context=incoming register = 36946811:[EMAIL PROTECTED]/1234 port=5060 bindaddr=0.0.0.0 srvlookup=yes ## extensions.conf [incoming] exten = _X.,Background(hello-world) ## sip debug (updated) --- SIP read from 87.54.25.114:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:87.54.25.114;ftag=as2dae750f;lr=on Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK9588.5b1985c.0 Via: SIP/2.0/UDP 87.54.25.116:5060;branch=z9hG4bK3fdd7e37;rport=5060 From: 23864098 sip:[EMAIL PROTECTED];tag=as2dae750f To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: no Max-Forwards: 16 Date: Sat, 26 Apr 2008 15:44:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 215 v=0 o=root 19760 19760 IN IP4 87.54.25.116 s=session c=IN IP4 87.54.25.116 t=0 0 m=audio 4 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - - --- (15 headers 10 lines) --- Sending to 87.54.25.114 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found peer 'musimi' --- Reliably Transmitting (NAT) to 87.54.25.114:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK9588.5b1985c.0;received=87.54.25.114 Via: SIP/2.0/UDP 87.54.25.116:5060;branch=z9hG4bK3fdd7e37;rport=5060 From: 23864098 sip:[EMAIL PROTECTED];tag=as2dae750f To: sip:[EMAIL PROTECTED];tag=as3fc8c57f Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5dc98c57 Content-Length: 0 Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) --- SIP read from 87.54.25.114:5060 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK9588.5b1985c.0 From: 23864098 sip:[EMAIL PROTECTED];tag=as2dae750f Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as3fc8c57f CSeq: 102 ACK Content-Length: 0 - --- (7 headers 0 lines) --- ___ -- Bandwidth and Colocation Provided 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] Channel variable settings
Hi all,
we are running Asterisk SVN-branch-1.4-r114299 and face following
problem: we have a main extension (102) and other extensions (104 and 107)
When extension 104 is calling extension 107 and 107 is on the phone, the
caller is parked for 5 seconds (Park-And-Announce) and the going back to
the called extension (busyNUMBER). After 6 loops in busy state, the call
has to be forwarded to main extension which is 102. Everything is
working perfectly with one call is this loop. If a second call enter
this loop, the busyNUMBER is replaced by the last asked extension, even
for calls already in the loop!
I was expecting that one call -here SIP channel- is one channel so
variables and their values are independent of the concurrent calls in
this channel. the busyNUMBER is _not_defined as a global variable and is
unique in the dialplan.
Is this behaviour normal or related to ParkAndAnnounce? Another solution
would be to create an unique variable like (tested but it's not working):
exten = _X.,1,Set(busyNUMBER=${UNIQUEID})
exten = _X.,n,Set($[${busyNUMBER}]=${EXTEN})
Is something like this existing?
Thanks for your feedback. Relevant part of the dialplan is:
[dial-local]
exten = _X.,1,Set(GLOBAL(__DIALEDNUMBER)=${EXTEN})
exten = _X.,n,Set(busyNUMBER=${EXTEN})
exten = _X.,n,Set(VoiceMail=u)
exten = _X.,n,Set(StatusPrio=1)
exten = _X.,n,macro(rec,)
exten = _X.,n,GotoIf($[${DIALEDNUMBER} =
${CALLERID(number)}]?extendedVM)
exten =
_X.,n(BackFromBusy),Dial(SIP/${DIALEDNUMBER},${SIPTIMERING},${DIALOPT})
exten = _X.,n,Goto(onDialStatus-local,s-${DIALSTATUS},${StatusPrio})
exten = _X.,20(extendedVM),NoOp(User ${CALLERID(number)} enter extended
voicemail)
exten = _X.,n,StopMonitor
exten = _X.,n,VoiceMailMain()
exten = _X.,n,Hangup
exten = _70[1-9],1,ParkedCall(${EXTEN})
[onDialStatus-local]
exten = s-BUSY,1,Set(VoiceMail=b)
exten = s-BUSY,n,Set(CHANNEL(LANGUAGE)=fr)
exten = s-BUSY,n,Set(countParkedLoop=0)
exten = s-BUSY,n,Background(busy-pls-hold)
exten =
s-BUSY,n(AfterAnnounce),Set(countParkedLoop=$[${countParkedLoop}+1])
exten = s-BUSY,n,GotoIf($[${countParkedLoop} 7]?:ReturnToMainLoop)
exten = s-BUSY,n,ParkAndAnnounce(pbx-transfer:PARKED|5|SIP/${busyNUMBER}})
exten = s-BUSY,n,Set(StatusPrio=AfterAnnounce)
exten = s-BUSY,n,Goto(dial-local,${busyNUMBER},BackFromBusy)
exten = s-BUSY,n(ReturnToMainLoop),Goto(dial-local,102,1)
--
Daniel
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Re: [asterisk-users] Need comments on CRM development / Asterisk Customization
Kashif Naeem wrote: Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? As well and Sugar and vtiger (PHP apps) also take a look at ConcursiveSuite (formerly known as CentricCRM) http://www.concursive.com. It has a crappy licence but has good asterisk integration. It's a JSP (Tomcat) application. HTH Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need comments on CRM development / Asterisk Customization
Some times ago i've started these projects: https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker Now i am too busy to update them, but you can use the main logic of ccmanager and the flexibility of reportmaker (you can define your report via xml) to make your own statistic about queues. Bye On Sat, Apr 26, 2008 at 6:12 PM, Alan Lord [EMAIL PROTECTED] wrote: Kashif Naeem wrote: Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? As well and Sugar and vtiger (PHP apps) also take a look at ConcursiveSuite (formerly known as CentricCRM) http://www.concursive.com. It has a crappy licence but has good asterisk integration. It's a JSP (Tomcat) application. HTH Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] yum install for specific kernel, how? And zaptel on fedora core 8
Dear All; If i need to download the source for the kernel 2.6.23.1-42.fc8-i686 (i do not need the latest one, i need this kernel specifically), what the command to be used? Also, any one tried to run zaptel 1.4.10 + asterisk 1.4.19 on fedora core 8? I am facing the problem that zaptel is not going online when booting, I tried a lot of solutions but did not work. Any one has idea? Do I need specific kernel to be used to work with zaptel 1.4.10? Any help? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided 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] yum install for specific kernel, how? And zaptel on fedora core 8
I am facing the problem that zaptel is not going online when booting, most people run it from /etc/rc.local ... ___ -- Bandwidth and Colocation Provided 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] yum install for specific kernel, how? And zaptel on fedora core 8
Install kernel-version.src.rpm with the following command: rpm -Uvh kernel-version.src.rpm this is from http://fedoraproject.org/wiki/Docs/CustomKernel ubuntu is better/safer/faster/has 5 year of updates ... you name it. On Sat, Apr 26, 2008 at 4:31 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear All; If i need to download the source for the kernel 2.6.23.1-42.fc8-i686 (i do not need the latest one, i need this kernel specifically), what the command to be used? Also, any one tried to run zaptel 1.4.10 + asterisk 1.4.19 on fedora core 8? I am facing the problem that zaptel is not going online when booting, I tried a lot of solutions but did not work. Any one has idea? Do I need specific kernel to be used to work with zaptel 1.4.10? Any help? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided 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] yum install for specific kernel, how? And zaptel on fedora core 8
Arthur wrote: I am facing the problem that zaptel is not going online when booting, most people run it from /etc/rc.local ... I thought most people ran it from /etc/rc.d/init.d/zaptel ___ -- Bandwidth and Colocation Provided 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] yum install for specific kernel, how? And zaptel on fedora core 8
I thought most people ran it from /etc/rc.d/init.d/zaptel here is what README file says : Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. -- make make install # To install init scripts and config files: #make config -- if you don't run make config you will be using /etc/rc.local ___ -- Bandwidth and Colocation Provided 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] yum install for specific kernel, how? And zaptel on fedora core 8
On Sat, Apr 26, 2008 at 09:31:32AM -0700, bilal ghayyad wrote: Dear All; If i need to download the source for the kernel 2.6.23.1-42.fc8-i686 (i do not need the latest one, i need this kernel specifically), what the command to be used? When all else fails, read the manual :-) http://zaptel.tzafrir.org.il/#_kernel_source_headers Also, any one tried to run zaptel 1.4.10 + asterisk 1.4.19 on fedora core 8? I am facing the problem that zaptel is not going online when booting, I tried a lot of solutions but did not work. Any one has idea? Do I need specific kernel to be used to work with zaptel 1.4.10? As you'll probably be able to see from the output of 'rpm -ql zaptel', that package contains no kernel modules, source to build them or whatever. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yum install for specific kernel, how? And zaptel on fedora core 8
On Sat, Apr 26, 2008 at 04:45:54PM +, Arthur wrote: Install kernel-version.src.rpm with the following command: rpm -Uvh kernel-version.src.rpm this is from http://fedoraproject.org/wiki/Docs/CustomKernel ubuntu is better/safer/faster/has 5 year of updates ... you name it. And packages of zaptel that are so well-tested that the gutsy one simply failed to build zaptel-modules package. And the Hardy one has not fixed many latest security issues of Asterisk. The packages 'asterisk' and 'zaptel' are in the universe repository of Ubuntu. Which means: not officially supported. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yum install for specific kernel, how? And zaptel on fedora core 8
On Sat, 26 Apr 2008, Arthur wrote: ubuntu is better/safer/faster/has 5 year of updates ... you name it. By what criteria do you form the opinion that Ubuntu is better, safer, and faster? Or do you have facts? I'm not looking for a flame-fest, just if you have objective measurements to move your statement from opinion to fact, I'd love to hear it. Especially if they hold true across all platforms -- SOHO to call center, SIP to PRI, VIA to Xeon and beyond! Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided 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] yum install for specific kernel, how? And zaptel on fedora core 8
On Sat, Apr 26, 2008 at 04:47:25PM +, Arthur wrote: I am facing the problem that zaptel is not going online when booting, most people run it from /etc/rc.local ... Most people who don't understand what init.d script are do that. And later they reboot the sysetm unnecessarily because they can't properly restart Asterisk. Recall that Asterisk must be run after Zaptel. Ordering of init.d scripts is something that is easy to assure and is assured with the default asterisk and zaptel installations. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-dev] zap not coming online on fedora 8
[Moving to asterisk-users] On Sat, Apr 26, 2008 at 09:55:24AM -0700, bilal ghayyad wrote: Dear Steve; Thanks for your kindly reply. First of all, which kernel u used when u used fedora core 8? About me, I used: wget http://ftp.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz and: wget http://ftp.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz Please note that I used make config to have automatic loading when booting, so when booting I get the following: Starting zaptel loading zaptel farmwork .. OK Waiting for zap to come online -- FAILURE Error missing /dev/zap What is the output of: modinfo zaptel lsmod | grep ^zaptel cat /proc/zaptel/* My digium card is TDM401 with 2 fxs and 2 fxo card. 401? 410? And ofcourse i do not find tone on my fxs. Is it related to the automatic initialization script (that will be used when run make config) or to my kernel or to what? Any need to do any kind of modification? Or I have to use yum instead of wget? yum is for fetching and installing rpm packages from remote yum repositories (and more, I know, but not for a simple wget replacement) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] init scripts (was: Re: yum install for specific kernel, how? And zaptel on fedora core 8)
Tzafrir Cohen schrieb: Most people who don't understand what init.d script are do that. And later they reboot the sysetm unnecessarily because they can't properly restart Asterisk. Recall that Asterisk must be run after Zaptel. Ordering of init.d scripts is something that is easy to assure and is assured with the default asterisk and zaptel installations. Unfortunately asterisk's init.d script for Debian is neither compliant with Debian's rules for init scripts nor LSB compliant. And the integration of safe_asterisk doesn't always work as expected. :-) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
Steve Totaro [EMAIL PROTECTED] writes: My dual proc, dual core AMD boxen show as four procs. I guess the AMD architecture uses Hypertheading (or whatever the equivalent is for AMD, I assume Intel owns the rights to the name Hyperthreading). I think the more likely explanation is that two times two is four. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yum install for specific kernel, how? And zaptel on fedora core 8
By what criteria do you form the opinion that Ubuntu is better, safer, and faster? well I don't know what's your favourite but compared to fedora ... i guess it is. ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
On Sat, Apr 26, 2008 at 2:37 PM, Benny Amorsen [EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: My dual proc, dual core AMD boxen show as four procs. I guess the AMD architecture uses Hypertheading (or whatever the equivalent is for AMD, I assume Intel owns the rights to the name Hyperthreading). I think the more likely explanation is that two times two is four. /Benny But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2. Or is C2D not four cores? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yum install for specific kernel, how? And zaptel on fedora core 8
By what criteria its a like a car you've got to drive it to feel it.. so give it a shot ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
On Sat, 26 Apr 2008, Steve Totaro wrote: On Sat, Apr 26, 2008 at 2:37 PM, Benny Amorsen [EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: My dual proc, dual core AMD boxen show as four procs. I guess the AMD architecture uses Hypertheading (or whatever the equivalent is for AMD, I assume Intel owns the rights to the name Hyperthreading). I think the more likely explanation is that two times two is four. /Benny But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2. Or is C2D not four cores? Have a look at the 'flag's in /proc/cpuinfo If there's a 'ht' then it's a hyperthreading core. Otherwise its a 'real' processor, and if there are N on a chip, then well and good. Hyperthreading is like having an extra quarter of a processor, depending on what tasks you're doing. But some BIOSes can disable the HT part of a core, and a modern kernel ought to have hyperthreading support compiled in to make the best use of it. (Another reason I always custom compile kernels for my applications) As far as I'm aware, AMD hasn't made a hyperthreaded core, so it's real cores on the same chip. So if the dual proc, dual core unit above has 2 CPU chips, then it's 4 processors. Going back to Intel: Core 2 is Intels name for that particular family of processors. The 2 in the name does not indicate the number of cores! So you can have a Core 2 Solo - one core, Core 2 Duo - 2 cores and Core 2 Quad - 4 cores. Gory details: http://en.wikipedia.org/wiki/Hyperthread http://en.wikipedia.org/wiki/Intel_Core_2 Gordon ___ -- Bandwidth and Colocation Provided 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] init scripts (was: Re: yum install for specific kernel, how? And zaptel on fedora core 8)
On Sat, Apr 26, 2008 at 08:24:45PM +0200, Philipp Kempgen wrote: Tzafrir Cohen schrieb: Most people who don't understand what init.d script are do that. And later they reboot the sysetm unnecessarily because they can't properly restart Asterisk. Recall that Asterisk must be run after Zaptel. Ordering of init.d scripts is something that is easy to assure and is assured with the default asterisk and zaptel installations. Unfortunately asterisk's init.d script for Debian is neither compliant with Debian's rules for init scripts nor LSB compliant. And the integration of safe_asterisk doesn't always work as expected. :-) safe_asterisk is buggy. Period. There's a limit to how much we can fix it. As for LSB-compliance: this is true indeed for the packages in Etch. Fixed later. And anyway, in Etch Debian has not supported service dependencies. In Lenny it will. That said, if you use the packages zaptel and asterisk, zaptel will start before asterisk. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] init scripts
Tzafrir Cohen schrieb: On Sat, Apr 26, 2008 at 08:24:45PM +0200, Philipp Kempgen wrote: Tzafrir Cohen schrieb: Most people who don't understand what init.d script are do that. And later they reboot the sysetm unnecessarily because they can't properly restart Asterisk. Recall that Asterisk must be run after Zaptel. Ordering of init.d scripts is something that is easy to assure and is assured with the default asterisk and zaptel installations. Unfortunately asterisk's init.d script for Debian is neither compliant with Debian's rules for init scripts nor LSB compliant. And the integration of safe_asterisk doesn't always work as expected. :-) safe_asterisk is buggy. Period. There's a limit to how much we can fix it. Sure. The issues with safe_asterisk have been discussed several times (http://bugs.digium.com/view.php?id=9843 etc.) so I didn't mean to start over. Never mind. And anyway, in Etch Debian has not supported service dependencies. In Lenny it will. Right. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
No, a dual core processor has two cores. :) My Quad core shows four processors. Steve Totaro wrote: On Sat, Apr 26, 2008 at 2:37 PM, Benny Amorsen [EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: My dual proc, dual core AMD boxen show as four procs. I guess the AMD architecture uses Hypertheading (or whatever the equivalent is for AMD, I assume Intel owns the rights to the name Hyperthreading). I think the more likely explanation is that two times two is four. /Benny But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2. Or is C2D not four cores? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:48137bd6112601373216315! ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
Steve Totaro [EMAIL PROTECTED] writes: But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2. Or is C2D not four cores? D is for duo. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manual Wardialer
Does anyone have a script for manual wardialer for asterisk? not sure if wardialer is the correct term but basically I want to call X number say 555- through 555-0050 and be able to listen to each call and when I hang up or press a key it will dial the next number for me. I guess sort of like scanning an exchange but I want to be on the line and if possible complete / talk on certain calls. ___ -- Bandwidth and Colocation Provided 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] Manual Wardialer
if wardialer is the correct term this must be a predictive dialer ... which is simply a dialer that dials a list of numbers you supply to him ... then you need to configure if for when to pass the call to you when to hung up ... vicidial is a good project to start with ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
D is for Disturbing other poeple. On Sat, Apr 26, 2008 at 10:39 PM, Benny Amorsen [EMAIL PROTECTED][EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2. Or is C2D not four cores? D is for duo. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manual Wardialer
On Sat, Apr 26, 2008 at 06:41:44PM -0400, Andreas van dem Helge wrote: Does anyone have a script for manual wardialer for asterisk? not sure if wardialer is the correct term but basically I want to call X number say 555- through 555-0050 and be able to listen to each call and when I hang up or press a key it will dial the next number for me. I guess sort of like scanning an exchange but I want to be on the line and if possible complete / talk on certain calls. core show application Dial ... Options: ... g- Proceed with dialplan execution at the current extension if the destination channel hangs up ... H- Allow the calling party to hang up by hitting the '*' DTMF digit. ... So all that's left for you is to run a loop of 50 Dial-s in the dialplan. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manual Wardialer
Andreas van dem Helge wrote: Does anyone have a script for manual wardialer for asterisk? not sure if wardialer is the correct term but basically I want to call X number say 555- through 555-0050 and be able to listen to each call and when I hang up or press a key it will dial the next number for me. I guess sort of like scanning an exchange but I want to be on the line and if possible complete / talk on certain calls. I think this is more of a power dialer rather than a predictive dialer, which uses a predictive algorithm to pace calling based on current and history data. I think vicidial is capable of both types of dialing I think. -- Warm Regards, Lee ___ -- Bandwidth and Colocation Provided 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] Quality problems with ISDN PRI
Got it, sorry, nothing to see here... I thought the 2 and duo meant 2 x 2 as the name would imply (to me at least). I stopped keeping up on processors with the exception of exceptional hardware like the ARM, RISC, FPGA, and Itanium2. Thanks, Steve Totaro On Sat, Apr 26, 2008 at 7:17 PM, Arthur [EMAIL PROTECTED] wrote: D is for Disturbing other poeple. On Sat, Apr 26, 2008 at 10:39 PM, Benny Amorsen [EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2. Or is C2D not four cores? D is for duo. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided 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] Manual Wardialer
On Sat, Apr 26, 2008 at 10:27 PM, Lee Jenkins [EMAIL PROTECTED] wrote: Andreas van dem Helge wrote: Does anyone have a script for manual wardialer for asterisk? not sure if wardialer is the correct term but basically I want to call X number say 555- through 555-0050 and be able to listen to each call and when I hang up or press a key it will dial the next number for me. I guess sort of like scanning an exchange but I want to be on the line and if possible complete / talk on certain calls. I think this is more of a power dialer rather than a predictive dialer, which uses a predictive algorithm to pace calling based on current and history data. I think vicidial is capable of both types of dialing I think. -- Warm Regards, Lee I suppose you could write a pretty simple perl (or whatever script) that could continually check the channel status and if idle, create a .call file with NXXNXX (variable set in script) and then loop it to dial NXXNXX + 1 or monitor the AMI for the Hangup event. I think Vicidial is a huge overkill for your description of needs. You make it sound as if only one person would be dialing one number at a time and only pause on an answer and only for the duration connected. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
