Re: [asterisk-users] How can i switch to samba server omitting sshfs
On Sun, Aug 1, 2010 at 10:34 PM, Janu Mukherjee janu.mu...@gmail.com wrote: Hi all, I have the following problem. I want to Call -- Asterisk AGI Answer -- Create File - Copy File Asterisk -- Play File -- Finish Call For now we are using sshfs to map the directories. I now want to achieve this using samba server. I am new to this concept and please help in this regard and provide me suggestions to move ahead. Thanks in Advance, Jahnavi. -- _ -- 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 This seems a bit like asking the telephone company for a howto on changing from regular light switches to square decora style light switches. What distro is the server using? Start with installing samba ;) -- _ -- 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 Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Well, actually we are in contact with quite a number of call-centers that use the free version - a lot of times it's embedded call centers, like internal help-desks and such. One of the nicest things of * is that you would not buy an ACD module for a traditional pbx to support just a couple of users, but with * it's free. l. 2010/7/31 bruce bruce bruceb...@gmail.com 2 users. So, it's probably never used as a free version as probably there are no 2 seat call centers that can survive this economy. But, it should great for testing. On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 7/30/2010 5:49 AM, Lenz Emilitri wrote: QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. Oh really! I didn't know that! Very nice. What is considered a small call centre? Are we talking up to around 5 agents or something? Is there a limit on the number of queues? (I'm sure there is a page on the website that answers most of these questions, heh :)) Leif Madsen. -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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
Re: [asterisk-users] How can i switch to samba server omitting sshfs
On Monday 02 Aug 2010, Janu Mukherjee wrote: Hi all, I have the following problem. I want to Call -- Asterisk AGI Answer -- Create File - Copy File Asterisk -- Play File -- Finish Call For now we are using sshfs to map the directories. I now want to achieve this using samba server. I am new to this concept and please help in this regard and provide me suggestions to move ahead. Unless you specifically want to share files with Windows machines, you will probably find that NFS -- Network File System, Unix's native file sharing -- is a better option than Samba. You just need a line like this in /etc/exports on the server; /usr/share/stuff 192.168.0.0/16(rw,sync,no_subtree_check) /usr/share/stuff is the directory to be exported (shared) 192.168.0.0/16 is the network range allowed to connect to it (rw,sync,no_subtree_check) are generic, sane options and this in /etc/fstab on the client; 192.168.0.2:/usr/share/stuff /stuff nfs defaults 0 0 192.168.0.2 is the server IP /usr/share/stuff is the directory to be mounted /stuff is the mountpoint on the local machine nfs is the file system type defaults is a generic, sane option 0 0 means don't include this file system in fsck See also the manpages for exports and fstab respectively. Note that depending on which distribution you are running, you may need to install some packages to make this work; refer to your distribution's homepage. -- AJS -- _ -- 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] Any Free software that can connect to an Asterisk Server and Do video Conferencing?
Hi, Is there any Free software that can connect to an Asterisk Server and Do video Conferencing? or atleast one to one video chat? thanks --Siju -- _ -- 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 Gurus - What is your best Ast erisk Queue Analyzer and Asterisk Log Analyzer program out ther e?
If you're only running a 2 agent call center, you could also take a look at http://www.orderlyq.com/asteriskcallcenterstatistics.html - its also free for 2 agents Rob On Sat, 31 Jul 2010 15:31:56 -0400, bruce bruce bruceb...@gmail.com wrote: 2 users. So, it's probably never used as a free version as probably there are no 2 seat call centers that can survive this economy. But, it should great for testing. On Sat, Jul 31, 2010 at 10:28 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 7/30/2010 5:49 AM, Lenz Emilitri wrote: QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. Oh really! I didn't know that! Very nice. What is considered a "small" call centre? Are we talking up to around 5 agents or something? Is there a limit on the number of queues? (I'm sure there is a page on the website that answers most of these questions, heh :)) Leif Madsen. -- _ -- 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
[asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200
Of course i use Wireshark and i see T.38 traffic but it isn't clear to me why the fax fails every time. I would like to know if there are T.38 tools/plugins that analyze a .pcap file more thorough. In my setup the analogue fax device and the ATA are near each other at the customer site. The ATA connects (via T.38) to the Asterisk server which is in a datacentre with the Cisco PGW. The PGW is connected to the public telephone network. Has anyone experience with sending T.38 fax messages between Asterisk and a Cisco PGW?-- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
No, from sources version 0.64 it's working fine. On Sat, Jul 31, 2010 at 10:59 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Fri, Jul 30, 2010 at 07:15:00AM -0400, Fred Posner wrote: On Jul 30, 2010, at 5:04 AM, Andraž wrote: Ok, problem is another, when I run configure, it write this: checking for tds_version in -ltds... no configure: *** configure: *** The FreeTDS installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** without explicitly specifying --with-tds ODBC is not a good solution, only if I can change the names of CDR fields. Hmm what was the exact '--with-tds' option you used? Have you passed any explicit path? How can I repair the installlation? If the above: try just not passing any explicit path. Or maybe even skipping this option altogether. Also: Have you tried installing freetds from source? This suggestion does not make sense to me. The package 1.6.2 I have here builds cdr_tds just fine. The Ubuntu package likewise: http://packages.ubuntu.com/lucid/amd64/asterisk/filelist As you can see from its build dependencies: http://packages.ubuntu.com/source/lucid/asterisk It depends on libfreetds-dev Have you installed it? As a rule of thumb, using: aptitude build-dep asterisk might be a good start. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
[asterisk-users] Codec negotiation : expecting G726, getting G711a (alaw)
Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the SIP trace : INVITE sip:2...@192.168.1.150 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9 From: User sip:u...@192.168.1.150;tag=2383fb163ee6befa To: sip:2...@192.168.1.150 Contact: sip:u...@192.168.1.102:5062;transport=udp Supported: replaces, timer, path Proxy-Authorization: Digest username=user, realm=domain.be, algorithm=MD5, uri=sip:2...@192.168.1.150, nonce=1ae22736, response=c90d0d9bf1f3c2bbc020651a5b67b608 Call-ID: 8910dbc6f2d5f...@192.168.1.102 CSeq: 35396 INVITE *User-Agent: Grandstream GXP2010 1.2.1.4* Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 250 v=0 o=user 8000 8001 IN IP4 192.168.1.102 s=SIP Call c=IN IP4 192.168.1.102 t=0 0 m=audio 10126 RTP/AVP 2 8 101 a=sendrecv *a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000* a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 - [Aug 2 13:56:57] --- (14 headers 12 lines) --- [Aug 2 13:56:57] Sending to 192.168.1.102 : 5062 (NAT) [Aug 2 13:56:57] Using INVITE request as basis request - 8910dbc6f2d5f...@192.168.1.102 [Aug 2 13:56:57] Found user 'user' [Aug 2 13:56:57] Found RTP audio format 2 [Aug 2 13:56:57] Found RTP audio format 8 [Aug 2 13:56:57] Found RTP audio format 101 [Aug 2 13:56:57] Found audio description format G726-32 for ID 2 [Aug 2 13:56:57] Found audio description format PCMA for ID 8 [Aug 2 13:56:57] Found audio description format telephone-event for ID 101 *[Aug 2 13:56:57] Capabilities: us - 0x90a (gsm|alaw|g726|g729), peer - audio=0x808 (alaw|g726)/video=0x0 (nothing), combined - 0x808 (alaw|g726)* [Aug 2 13:56:57] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Aug 2 13:56:57] Peer audio RTP is at port 192.168.1.102:10126 [Aug 2 13:56:57] Looking for 20 in from-STERKEN (domain 192.168.1.150) [Aug 2 13:56:57] list_route: hop: sip:u...@192.168.1.102:5062;transport=udp [Aug 2 13:56:57] --- Transmitting (NAT) to 192.168.1.102:5062 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102 From: User sip:u...@192.168.1.150;tag=2383fb163ee6befa To: sip:2...@192.168.1.150 Call-ID: 8910dbc6f2d5f...@192.168.1.102 CSeq: 35396 INVITE User-Agent: my-asterisk-server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:2...@192.168.1.150 Content-Length: 0 - [Aug 2 13:56:57] --- (11 headers 0 lines) --- [Aug 2 13:56:57] SIP Response message for INCOMING dialog NOTIFY arrived [Aug 2 13:56:57] -- SIP/sterkendries2-0054 is ringing [Aug 2 13:56:57] --- Transmitting (NAT) to 192.168.1.102:5062 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102 From: User sip:u...@192.168.1.150;tag=2383fb163ee6befa To: sip:2...@192.168.1.150;tag=as655a8251 Call-ID: 8910dbc6f2d5f...@192.168.1.102 CSeq: 35396 INVITE *User-Agent: my-asterisk-server* Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:2...@192.168.1.150 Content-Length: 0 --- [Aug 2 13:57:00] Extension Changed 20[105002-blf] new state InUse for Notify User user [Aug 2 13:57:00] -- SIP/sterkendries2-0054 answered SIP/user-0053 [Aug 2 13:57:00] Audio is at 192.168.1.150 port 11500 [Aug 2 13:57:00] Adding codec 0x8 (alaw) to SDP [Aug 2 13:57:00] Adding codec 0x800 (g726) to SDP [Aug 2 13:57:00] Adding non-codec 0x1 (telephone-event) to SDP [Aug 2 13:57:00] --- Reliably Transmitting (NAT) to 192.168.1.102:5062 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:5062;branch=z9hG4bK1dc39a962389c1a9;received=192.168.1.102 From: User sip:u...@192.168.1.150;tag=2383fb163ee6befa To: sip:2...@192.168.1.150;tag=as655a8251 Call-ID: 8910dbc6f2d5f...@192.168.1.102 CSeq: 35396 INVITE *User-Agent: my-asterisk-server* Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:2...@192.168.1.150 Content-Type: application/sdp Content-Length: 267 v=0 o=root 1947 1947 IN IP4 192.168.1.150 s=session c=IN IP4 192.168.1.150 t=0 0 m=audio 11500 RTP/AVP 8 2 101 *a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv - [Aug 2 13:57:00] --- (11 headers 0 lines) --- [Aug 2 13:57:00] SIP Response message for INCOMING dialog NOTIFY arrived [Aug 2 13:57:00] --- SIP read from
[asterisk-users] Asterisk and TV media server
Hello, I would like to know whether there is a way to associate a TV media server with Asterisk. Is it possible to access TV Chanels in the Telephone Sets. Anybody have any tips or documents related to this please let me know. Thanks -- _ -- 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] how to place a call on hold and play music on holdusing agi
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Janu Mukherjee Subject: [asterisk-users] how to place a call on hold and play music on holdusing agi I want to originate a call using asterisk agi. I could this. I now want to place this call on hold and play music on hold and after some time i want to retrieve the call. Can i do this using AGI?Please help me in this regard. This is not truly putting the call on hold but will work the same way; set up this context; [dummy-hold] Exten = s,1,Waitexten(60,m) Then make your call go to dummy-hold and you will be on-hold with music for 1 minute and then you can go back to the dialplan or put a return on the context and make it a Gosub. -- _ -- 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] mapping of disconnect reasons
Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 “Payment Required” from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is resulting call status “CONGESTION”) but will not do so for all normal terminations (16, Normal Termnation, 17 Busy, 18 No Answer). Thanks, Harel -- _ -- 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] asterisk compatible cards?
hello, i just subscribed to this list, i discovered asterisk and i would like to try it at home on my personal pc. the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit guarranted connection and runs a gentoo linux. i search about digium products but i can't find them in my area on any shops, i was wondering if good people here could recommend some PCI or PCIex cards for a beginner to play with one telefonic line (which i will install it soon via provider) thanks! -- _ -- 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 compatible cards?
On Mon, 2 Aug 2010, Daniel Petre wrote: hello, i just subscribed to this list, i discovered asterisk and i would like to try it at home on my personal pc. the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit guarranted connection and runs a gentoo linux. i search about digium products but i can't find them in my area on any shops, i was wondering if good people here could recommend some PCI or PCIex cards for a beginner to play with one telefonic line (which i will install it soon via provider) If you really can't get digium cards, then look on ebay for x100p cards - you might get lucky... Failing that, OpenVox have some compatable cards - you might find an importer locally who deals in them. Gordon -- _ -- 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] R: asterisk compatible cards?
Hi Daniel, have a look at this page, maybe it will help you find a reseller: http://www.voip-info.org/wiki/view/Asterisk+Consultants+Romania . Best Regards, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Daniel Petre Inviato: lunedì 2 agosto 2010 15:36 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [asterisk-users] asterisk compatible cards? hello, i just subscribed to this list, i discovered asterisk and i would like to try it at home on my personal pc. the computer is a p4 at 3 ghz with 2 gb ram and 80 gb hdd, a 1 Mbit guarranted connection and runs a gentoo linux. i search about digium products but i can't find them in my area on any shops, i was wondering if good people here could recommend some PCI or PCIex cards for a beginner to play with one telefonic line (which i will install it soon via provider) thanks! -- _ -- 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] Any Free software that can connect to an Asterisk Server and Do video Conferencing?
On 08/02/2010 02:34 AM, Siju George wrote: Hi, Is there any Free software that can connect to an Asterisk Server and Do video Conferencing? or atleast one to one video chat? One to one video chat is already supported by Asterisk, using SIP or H.323 video phones. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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 and TV media server
On Mon, Aug 2, 2010 at 5:37 AM, Tino t...@sparksupport.com wrote: Hello, I would like to know whether there is a way to associate a TV media server with Asterisk. Is it possible to access TV Chanels in the Telephone Sets. Anybody have any tips or documents related to this please let me know. Thanks -- _ -- 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 That idea could go two ways, dial a number and get audio, or dial a number with a video phone and watch the channel. The video phone idea sounds like it'd be neat to use. -- _ -- 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 and TV media server
Sorry, I am a newbie to this concept. Can you please briefly explain how it is possible to watch TV channels using a video phone by just dialing a number. Is there any website links that you can share with me on this subject ? . Thanks for your interest in this matter. On Mon, Aug 2, 2010 at 8:54 PM, Kyle Kienapfel doctor.w...@gmail.comwrote: On Mon, Aug 2, 2010 at 5:37 AM, Tino t...@sparksupport.com wrote: Hello, I would like to know whether there is a way to associate a TV media server with Asterisk. Is it possible to access TV Chanels in the Telephone Sets. Anybody have any tips or documents related to this please let me know. Thanks -- _ -- 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 That idea could go two ways, dial a number and get audio, or dial a number with a video phone and watch the channel. The video phone idea sounds like it'd be neat to use. -- _ -- 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] Asterisk and TV media server
On Mon, Aug 2, 2010 at 8:37 AM, Tino t...@sparksupport.com wrote: Anybody have any tips or documents related to this please let me know. http://www.youtube.com/watch?v=3h6-PSpD-Oc -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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
[asterisk-users] Stupid Macro question
Hi all,
I am just trying to implement DUNDi-Routing like described here
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
and have a most probably stupid question:
My config is exactly like described except that instead of
exten = _91NXXNXX,1,Macro(dundi-priv,${EXTEN:1})
exten = _91NXXNXX,2,Dial(Zap/g1/${EXTEN:1}) ; This is fall through example
to a PSTN such a as PRI
I have
exten = _X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
to check every dialed number via DUNDi and if not reachable via DUNDi, call via
PSTN.
Now my problem is, that after hanging up a call, the call is instantly
re-established using the h-extension which is almost a loop.
I am sure this is a stupid question, but what am I doing wrong?
Thanks for advice
Oliver
--
_
-- 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] IAX softphone
Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo. -- _ -- 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] Stupid Macro question
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
Subject: [asterisk-users] Stupid Macro question
Hi all,
I have
exten = _X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
Now my problem is, that after hanging up a call, the call is instantly
re-established using the h-extension which is almost a loop.
I am sure this is a stupid question, but what am I doing wrong?
Thanks for advice
Oliver
This might fix you up
exten = _X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
exten = _x_NOANSWER,1,Dial(Zap/g1/${EXTEN:1}) ;
This way the Zap call only occurs on a DUNDI noanswer.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX softphone
On 02/08/10 17:35, Ronaldo Zacarias Afonso wrote: Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. This one is great on Ubuntu/Linux. http://www.sflphone.org/ Unfortunately I know not about Windows though, I never use it. Cheers Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisknow
Is a mail server built in in asterisk now Like in elastix -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stupid Macro question
Hi all,
I have
exten =_X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
Now my problem is, thatafter hanging up a call, the call is instantly
re-established using theh-extension which is almost a loop.
I am sure this is astupid question, but what am I doing wrong?
Thanks for advice
Oliver
This might fix you up
exten =_X.,1,Macro(dundi-priv,${EXTEN})
exten = _X.,2,DIAL(CAPI/contr1/${EXTEN})
exten = _x_NOANSWER,1,Dial(Zap/g1/${EXTEN:1}) ;
This way the Zap callonly occurs on a DUNDI noanswer.
--
Thanks, but that is not the problem, DUNDi is answering and is forwarding the
call to the remote box.
That works fine.
But immediately after hanging up the call by the client registered on the
remote box the call is re-established
using the h-extension.
This is my problem.
--
_
-- 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] IAX softphone
I use http://www.voixphone.com/ On Mon, Aug 2, 2010 at 9:41 PM, Alan Lord (News) alansli...@gmail.comwrote: On 02/08/10 17:35, Ronaldo Zacarias Afonso wrote: Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. This one is great on Ubuntu/Linux. http://www.sflphone.org/ Unfortunately I know not about Windows though, I never use it. Cheers Al -- The Open Learning Centre http://www.theopenlearningcentre.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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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
Re: [asterisk-users] asterisknow
On Mon, Aug 2, 2010 at 9:56 AM, mattias m...@mjw.se wrote: Is a mail server built in in asterisk now Like in elastix -- _ -- 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 Asterisknow is a linux distribution? You could probably add one easily if its missing. It's got a mailing list which is cool http://lists.digium.com/mailman/listinfo/asterisknow -- _ -- 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] What do you use for Invoicing?
Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Thanks -- _ -- 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] What do you use for Invoicing?
On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Its partially open source (you get the source to everything but the financial routines), and it runs on Unix rather than Windows, though you do have a web interface. Checkout BillMax: www.billmax.com They have some extensions that create PDF invoices in telecom style. Its pretty powerful otherwise for doing any kind of recurring billing. I wrote the initial version, but I am not associated with the company anymore. j Thanks -- _ -- 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] IAX softphone
On Mon, 2 Aug 2010, Ronaldo Zacarias Afonso wrote: Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. www.Zoiper.com Gordon -- _ -- 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] What do you use for Invoicing?
On Mon, 2 Aug 2010, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. I generated invoices with PHP code - it uses a LaTeX template which it fills in the gaps, then feeds it through LaTeX and dvi2pdf to generate PDFs. Bit of a geek solution though. Gordon -- _ -- 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] FAX Options
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Subject: [asterisk-users] FAX Options Is FAXing with Asterisk a practical option ? Or is it better just to use a plain fax connected to an FXS and just switch with Asterisk. I specifically wanted to know if there was any experience using just the fax scanner to send faxes and receive them via asterisk and the to e-mail. My idea was to take my old fax connect it to an FXS port and send faxes with the fax machine (using the fax mainly as a scanner), but receive them through our existing FXO jack that is connected to the PSTN. the scheme would be something like: PSTN -- FXO - | |Asterisk | FAX -- FXS - I'm using Asterisk 1.4.26.2 on FreeBSD 8.0 TIA, Alejandro Imass IMO, as long as you're using PSTN and nothing fancy like T.38, Asterisk is a solid fax send/receive option. -- _ -- 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 Options
Hi, Is FAXing with Asterisk a practical option ? Or is it better just to use a plain fax connected to an FXS and just switch with Asterisk. I specifically wanted to know if there was any experience using just the fax scanner to send faxes and receive them via asterisk and the to e-mail. My idea was to take my old fax connect it to an FXS port and send faxes with the fax machine (using the fax mainly as a scanner), but receive them through our existing FXO jack that is connected to the PSTN. the scheme would be something like: PSTN -- FXO - | |Asterisk | FAX -- FXS - I'm using Asterisk 1.4.26.2 on FreeBSD 8.0 TIA, Alejandro Imass -- _ -- 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] Whither app_nv_faxdetect
Anyone know where the sources for app_nv_faxdetect officially live? I couldn't turn them up on a web search, just patched versions for 1.4, etc. Thanks. -- _ -- 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] fail2ban does not work for my asterisk installation
Thanks for your reply. My configuration is correct. It works with ssh: many attacks have been stopped. Also, the config has worked for asterisk one time: I have seen that in the fail2ban.log file. -- _ -- 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] Femtocell to VoIP?
Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -- _ -- 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] Femtocell to VoIP?
On Mon, Aug 2, 2010 at 3:36 PM, Matt mhop...@gmail.com wrote: Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? I have not, but I have had great luck with OpenBTS. Thanks, Steve T -- _ -- 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] Caller ID issue
Hi list,
I'm having a problem with CallerID names not showing up when calls come
in. I have dialplan code to store the callerid(name) away and it is
blank (null). However, the voicemail variable ${VM_CALLERID} has the
name field populated. For example, here is some of the dialplan code:
2. Set(CALLER_ID_INFO_ALL=${CALLERID(all)})
3. Set(CALLER_ID_INFO_NAME=${CALLERID(name)})
4. Set(CALLER_ID_INFO_NUM=${CALLERID(num)})
5. Set(CALLER_ID_INFO_ANI=${CALLERID(ANI)})
6. Set(CALLER_ID_INFO_DNID=${CALLERID(DNID)})
Which yields this at the CLI:
-- Executing [3...@from_outside:2] Set(DAHDI/1-1,
CALLER_ID_INFO_ALL= 2565551212) in new stack
-- Executing [3...@from_outside:3] Set(DAHDI/1-1,
CALLER_ID_INFO_NAME=) in new stack
-- Executing [3...@from_outside:4] Set(DAHDI/1-1,
CALLER_ID_INFO_NUM=2565551212) in new stack
-- Executing [3...@from_outside:5] Set(DAHDI/1-1,
CALLER_ID_INFO_ANI=2565551212) in new stack
Note the first line should have the name field with the number, but does
not.
HOWEVER the voicemail notification contains:
Just wanted to let you know you were just left a 0:04 long message
(number 1) in mailbox 3703 from SMITH CASSIUS 2565551212
So - I know the NAME field is getting into the system, but it's not
showing up on the phones (and with telemarketers, that annoys my
users).
I'm using Asterisk 1.6.2.9, DAHDI 2.3.0
I have added callerid=asreceived to chan_dahdi.conf for my inbound
trunks, and shrinkcallerid=no to my sip.conf. (without effect)
Any ideas?
THANKS
Cassius
--
_
-- 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] Femtocell to VoIP?
On Mon, Aug 2, 2010 at 3:53 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Mon, Aug 2, 2010 at 3:36 PM, Matt mhop...@gmail.com wrote: Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? I have not, but I have had great luck with OpenBTS. Steve, Thanks... my only problem is right now OpenBTS is not cost effective to deploy to customers. -- _ -- 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] alaw.h in app_meetme.c
Hi Group, short question. is it possible to use #include asterisk/alaw.h instead of #include asterisk/ulaw.h in app_meetme.c or is ulaw required in meetme? thanx for the answer. Daniel -- _ -- 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] FAX Options
On Mon, Aug 2, 2010 at 3:03 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Subject: [asterisk-users] FAX Options [...] TIA, Alejandro Imass IMO, as long as you're using PSTN and nothing fancy like T.38, Asterisk is a solid fax send/receive option. Could you recommend a good starting point? Like a faxing with Asterisk how-to... -- _ -- 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] FAX Options
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass Subject: Re: [asterisk-users] FAX Options Could you recommend a good starting point? Like a faxing with Asterisk how-to... I would personally get the Free Fax for Asterisk. It is well documented and as long as you are using 1 line for fax, pretty much does what you want in as close to plug-and-play as anything else Asterisk does. -- _ -- 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] Caller ID issue
On Mon, Aug 2, 2010 at 2:56 PM, Cassius Smith cass...@cassius.org wrote: Any ideas? THANKS Cassius Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. -- Thanks, --Warren Selby http://www.selbytech.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
Re: [asterisk-users] Caller ID issue
Thanks Warren. That fixed it. I am using T1's and didn't think the spill would take that long. Ciao, Cassius Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. -- _ -- 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] Caller ID issue
Un-top-posting... On Mon, 2 Aug 2010, Cassius Smith wrote: I'm having a problem with CallerID names not showing up when calls come in. On Mon, 2 Aug 2010, Warren Selby wrote: Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. On Mon, 2 Aug 2010, Cassius Smith wrote: Thanks Warren. That fixed it. I am using T1's and didn't think the spill would take that long. PRI no, EM yes. Using answer(2000) should also work. Can you try it and reply with your results? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Femtocell to VoIP?
On Mon, Aug 02, 2010 at 03:36:59PM -0400, Matt wrote: Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? Most people seem to be concentrating on 3G femtocells (there are various companies making designs based on picoChip soft radios). OpenBTS can be used (and there have been some successful quite large installations). Hay Systems were meant to be producing a 2G (GSM/GPRS) femtocell, but they seem to have gone quiet. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- 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] FAX Options
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Monday, August 02, 2010 9:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] FAX Options
Hi,
Is FAXing with Asterisk a practical option ? Or is it better just to
use a plain fax connected to an FXS and just switch with Asterisk. I
specifically wanted to know if there was any experience using just the
fax scanner to send faxes and receive them via asterisk and the to
e-mail. My idea was to take my old fax connect it to an FXS port and
send faxes with the fax machine (using the fax mainly as a scanner),
but receive them through our existing FXO jack that is connected to
the PSTN. the scheme would be something like:
PSTN -- FXO -
|
|Asterisk
|
FAX -- FXS -
I'm using Asterisk 1.4.26.2 on FreeBSD 8.0
Here we have the following setup, could you say if that is acceptable for
you?
Outgoing fax:
Fax - Linksys pap2t (sip, no t38, for settings see
http://www.provu.co.uk/pdf/sipura/ip_faxing_sipura_linksys.pdf) - asterisk
- sip trunk provider (this could also be some sip - pstn solution I guess)
Incoming fax:
Sip trunk provider - asterisk - email
For the incoming fax I use a separate context, below I've listed an example:
exten = 1000,1,Answer
exten =
1000,2,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}_${EPOCH}_client.
tif)
exten = 1000,3,Set(CLID=${CALLERID(num)})
exten = 1000,4,Set(EMAIL=email address)
exten = 1000,5,Set(TRADENAME=tradename (used in the email))
exten = 1000,6,Wait(3)
exten = 1000,7,ReceiveFax(${FAXFILE})
exten = 1000,8,Hangup
exten = h,1,System(/usr/bin/php /etc/scripts/fax2mail.php ${FAXFILE}
${CLID} ${EMAIL} ${TRADENAME})
fax2mail.php (tiff2pdf and phpmailer are required):
?php
$faxfile = $_SERVER[argv][1];
$callerid = $_SERVER[argv][2];
$email = $_SERVER[argv][3];
shell_exec(/usr/bin/tiff2pdf
-o/var/spool/asterisk/fax/.$callerid..pdf .$faxfile);
$bijlage = /var/spool/asterisk/fax/.$callerid..pdf;
switch ( $_SERVER[argv][4] ) {
case trade: $tradename = 'our trademark'; $from =
'f...@domain.tld';$fromname = $tradename.' - Fax system'; break;
default: $tradename = 'our trademark'; $from =
'f...@domain.tld';$fromname = $tradename.' - Fax system'; break;
}
require(/etc/scripts/class.phpmailer.php);
$mail = new PHPMailer();
// $mail-IsMail(); // telling the class to use Mail functie
van PHP
$mail-IsSMTP(); // telling the class to use SMTP
$mail-Host = ; // SMTP server
$mail-SMTPAuth = true; // turn on SMTP authentication
$mail-Username = ; // SMTP username
$mail-Password = ; // SMTP password
$mail-From = $from;
$mail-AddAddress($email);
$mail-FromName = $fromname;
$mail-AddAttachment($bijlage);
$mail-Subject = Received fax from .$callerid;
$mail-AddReplyTo = $email;
$mail-IsHTML(false);
$mail-Body = email body;
$mail-Send();
//shell_exec(/bin/rm /var/spool/asterisk/fax/.$callerid..pdf
.$faxfile);
?
I agree that it isn't a beautiful solution, however it works. Sending a fax
directly with asterisk is probably also possible (I didn't test it).
Asterisk version: 1.6.2.6 (yes I know that I should update)
Regards, Mark
TIA,
Alejandro Imass
--
_
-- 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] What TERMINAL software do you use for MS Windows platform and WHY?
On 30/06/10 1:53 AM, bruce bruce wrote: Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etcbut it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the IP address as the title of the window would help a lot if you have many different servers opened at the same time. Can you please weigh in and tell me what your favorite terminal software is and why? Late response, and I don't use Windows any more, but SecureCRT with tabbed SSH windows and buttons which can be set up for things like nano /etc/asterisk/extensions.conf make life pretty simple. On Mac I now use iTerm (similar thing). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Good script to make appointment?
On 16/07/10 4:40 AM, Gilles wrote: Hello I'd like to write a script that would make it easier for people to call in, listen to the IVR, and make an appointment (eg. When? ASAP? A given day? - Morning? Afternon, etc.) I assume I'm not the first one to try and write this type of IVR, so would appreciate any feedback on writing this. http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+Wake-Up+Call+PHP -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] FAX Options
On Mon, Aug 2, 2010 at 7:26 PM, Mark Scholten m...@streamservice.nl wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Alejandro Imass Sent: Monday, August 02, 2010 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FAX Options [...] Here we have the following setup, could you say if that is acceptable for you? Thanks! Looks very much like what we're looking for... I am sure it will work for us but we have 1.4 let me test some stuff and get back to you here Outgoing fax: Fax - Linksys pap2t (sip, no t38, for settings see http://www.provu.co.uk/pdf/sipura/ip_faxing_sipura_linksys.pdf) - asterisk - sip trunk provider (this could also be some sip - pstn solution I guess) Thanks again! I will test this by the end of the week and post my results here to follow-up and close the thread. Alejandro Imass -- _ -- 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] Asterisk 1.6 and PrivacyManager with SIP
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with Asterisk 1.4 is the PrivacyManager. However, this was not
straightforward, because anonymous SIP calls arrive with
${CALLERID(num)} = anonymous, instead of being blank. So, to get it
to work I added the first three rules to the following:
exten = jaap,1,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jaap,n(true),Set(CALLERID(num)=)
exten = jaap,n(false),NoOp()
exten = jaap,n,PrivacyManager()
exten = jaap,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jaap,n,Dial(SIP/1000,20,w)
exten = jaap,n,Hangup()
exten = jaap,n(bad),Playback(im-sorry)
exten = jaap,n,Playback(vm-goodbye)
exten = jaap,n,Hangup()
Unfortunately, this no longer seems to work with Asterisk 1.6: the
second rule is still executed, but for some reason the PrivacyManager
always decides that the caller ID is present anyway.
Should I be doing this differently now, or is something else wrong?
Thanks,
Jaap
--
_
-- 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] Caller ID issue
I am using T1's and didn't think the spill would take that long. PRI no, EM yes. Some PRI take that long too because the telco sends the name in a followup message, not in the initial call setup. -- _ -- 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] chinaroby fxo card - never heard of them
Hello. I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue. Thanks. -- _ -- 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 1.6 and PrivacyManager with SIP
Try removing the quotes in your n(true) priority.
Thanks,
--Warren Selby
On Aug 2, 2010, at 7:40 PM, Jaap Winius jwin...@umrk.nl wrote:
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with Asterisk 1.4 is the PrivacyManager. However, this was not
straightforward, because anonymous SIP calls arrive with
${CALLERID(num)} = anonymous, instead of being blank. So, to get it
to work I added the first three rules to the following:
exten = jaap,1,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jaap,n(true),Set(CALLERID(num)=)
exten = jaap,n(false),NoOp()
exten = jaap,n,PrivacyManager()
exten = jaap,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jaap,n,Dial(SIP/1000,20,w)
exten = jaap,n,Hangup()
exten = jaap,n(bad),Playback(im-sorry)
exten = jaap,n,Playback(vm-goodbye)
exten = jaap,n,Hangup()
Unfortunately, this no longer seems to work with Asterisk 1.6: the
second rule is still executed, but for some reason the PrivacyManager
always decides that the caller ID is present anyway.
Should I be doing this differently now, or is something else wrong?
Thanks,
Jaap
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Options
- Mark Scholten m...@streamservice.nl wrote:
Here we have the following setup, could you say if that is acceptable
for
you?
Outgoing fax:
Fax - Linksys pap2t (sip, no t38, for settings see
http://www.provu.co.uk/pdf/sipura/ip_faxing_sipura_linksys.pdf) -
asterisk
- sip trunk provider (this could also be some sip - pstn solution I
guess)
Incoming fax:
Sip trunk provider - asterisk - email
For the incoming fax I use a separate context, below I've listed an
example:
exten = 1000,1,Answer
exten =
1000,2,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}_${EPOCH}_client.
tif)
exten = 1000,3,Set(CLID=${CALLERID(num)})
exten = 1000,4,Set(EMAIL=email address)
exten = 1000,5,Set(TRADENAME=tradename (used in the email))
exten = 1000,6,Wait(3)
exten = 1000,7,ReceiveFax(${FAXFILE})
exten = 1000,8,Hangup
And this works for you? Fax over SIP is typically horrible with success rates
around 50%. T.38 improves this and you may be able to bring your success rates
up to the low 90%'s. That still isn't good enough for me, and I would imagine
any business that relies on proper fax connectivity.
Steve Underwood has a fantastic write up of issues surrounding fax over voip at
his site: http://www.soft-switch.org/foip.html
--Tim
--
_
-- 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] fail2ban does not work for my asterisk installation
On Mon, Aug 2, 2010 at 12:15 PM, mosbah abdelkader
mosbah.abdelka...@gmail.com wrote:
Thanks for your reply.
My configuration is correct. It works with ssh: many attacks have been
stopped. Also, the config has worked for asterisk one time: I have seen that
in the fail2ban.log file.
--
_
-- 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
p...@prometheus:/var/log/asterisk# sudo cat /etc/fail2ban/filter.d/asterisk.conf
# http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
[Definition]
#_daemon = asterisk
# Option: failregex
# Notes.: regex to match the password failures messages in the logfile. The
# host must be matched by a group named host. The tag HOST can
# be used for standard IP/hostname matching and is only an alias for
# (?:::f{4,6}:)?(?Phost\S+)
# Values: TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Wrong password
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
No matching peer found
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Username/auth name mismatch
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
Device does not match ACL
NOTICE.* HOST failed to authenticate as '.*'$
NOTICE.* .*: No registration for peer '.*' \(from HOST\)
NOTICE.* .*: Host HOST failed MD5 authentication for '.*' (.*)
NOTICE.* .*: Failed to authenticate user .*@HOST.*
NOTICE.* .*: Registration from '.*' failed for 'HOST' -
ACL error \(permit/deny\)
# Option: ignoreregex
# Notes.: regex to ignore. If this regex matches, the line is ignored.
# Values: TEXT
#
ignoreregex =
p...@prometheus:/var/log/asterisk# sudo
I don't see slashes in front of the brackets on what you posted to the
mailing list. I'm posting my config to see if the mailing list mangles
it or not.
--
_
-- 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 1.6 and PrivacyManager with SIP
Quoting Warren Selby wcse...@selbytech.com:
Try removing the quotes in your n(true) priority.
From FAILED? That makes no difference: with or without the quotes,
the result is always 0, which leads in the Dial() rule being executed.
Actually, though, that's not even relevant, because before Asterisk
even reaches that rule, the CLI shows that the result from the
PrivacyManager is:
-- CallerID Present: Skipping
PrivacyManager is simply failing to determine that the incoming SIP
calls are anonymous.
Actually, could it be that the second rule of my code, with the Set()
command, is simply not working with Asterisk 1.6? Let me try that
without the empty set of quotes after the equals sign...
Yes, that was it -- it's working again! Here's what it looks like now:
exten = jaap,1,GotoIf($[${CALLERID(num)}=anonymous]?true:false)
exten = jaap,n(true),Set(CALLERID(num)=)
exten = jaap,n(false),NoOp()
exten = jaap,n,PrivacyManager(3,10)
exten = jaap,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jaap,n,Dial(SIP/1000,20,w)
exten = jaap,n,Hangup()
exten = jaap,n(bad),Playback(im-sorry)
exten = jaap,n,Playback(vm-goodbye)
exten = jaap,n,Hangup()
Rule five now has both ${PRIVACYMGRSTATUS} and FAILED without quotes,
but that actually did not make any difference. Two things actually
fixed the problem. The first and most important was removing the pair
of empty quotes from rule two -- otherwise the caller ID is no longer
regarded as empty. Second is the addition of 3,10 as options to the
PrivacyManager application in rule four. Those are supposed to be the
defaults, but without them the PrivacyManager fails to recognize a
ten-digit phone number as being sufficient. I consider that a bug.
Cheers,
Jaap
--
_
-- 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] What do you use for Invoicing?
Maybe good but the first look brought me to a Pay version. Doesn't satisfy the opensource condition. thanks, On Mon, Aug 2, 2010 at 2:39 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Its partially open source (you get the source to everything but the financial routines), and it runs on Unix rather than Windows, though you do have a web interface. Checkout BillMax: www.billmax.com They have some extensions that create PDF invoices in telecom style. Its pretty powerful otherwise for doing any kind of recurring billing. I wrote the initial version, but I am not associated with the company anymore. j Thanks -- _ -- 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 -- _ -- 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] What do you use for Invoicing?
Sorry, I am not familiar with them. Wondering if any full package system out there does the job. Thanks On Mon, Aug 2, 2010 at 2:55 PM, Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net wrote: On Mon, 2 Aug 2010, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. I generated invoices with PHP code - it uses a LaTeX template which it fills in the gaps, then feeds it through LaTeX and dvi2pdf to generate PDFs. Bit of a geek solution though. Gordon -- _ -- 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] What do you use for Invoicing?
You forgot to say for free --Don _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Monday, August 02, 2010 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What do you use for Invoicing? Sorry, I am not familiar with them. Wondering if any full package system out there does the job. Thanks On Mon, Aug 2, 2010 at 2:55 PM, Gordon Henderson gordon+aster...@drogon.net mailto:gordon%2baster...@drogon.net wrote: On Mon, 2 Aug 2010, bruce bruce wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. I generated invoices with PHP code - it uses a LaTeX template which it fills in the gaps, then feeds it through LaTeX and dvi2pdf to generate PDFs. Bit of a geek solution though. Gordon -- _ -- 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] What do you use for Invoicing?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Tuesday, 3 August 2010 1:58 PM To: j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What do you use for Invoicing? Maybe good but the first look brought me to a Pay version. Doesn't satisfy the opensource condition. thanks, Open Source software does not necessarily mean free software. Nick. -- _ -- 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] chinaroby fxo card - never heard of them
hi, I am using this card and IP phone about 6 months. There is no issues at all. Installation procedures are same as Digium analog card. Hope it helps, Ashik On Tue, Aug 3, 2010 at 6:28 AM, Landy Landy landysacco...@yahoo.com wrote: Hello. I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue. Thanks. -- _ -- 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
