Re: [Asterisk-Users] RE: Predictive Dialer
Adam Vocks wrote: OK, so apparently no one is using GnuDialer, is anyone out there using any other predictive dialers on asterisk? Thank you, Adam Vocks *From:* Adam Vocks *Sent:* Thursday, March 09, 2006 12:41 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Predictive Dialer Hello all, I have a client interested in GnuDialer. My question is: Is there anyone on this list who has been using GnuDialer and I was wondering if you would be willing to share your experiences with it. Thank You Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I am using VCIDialer for testing purposes.. and work fine... 70 concurrent calls, a little heavy to install regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 + Sonicwall
Rich Adamson wrote: Hi all, I currently have an Asterisk test server behind a TZ170 Sonicwall firewall / NAT box, with several DIDs. I've found that inbound IAX2 calls don't work reliably (i.e., I get a busy tone) unless I enable Use Consistent NAT in the Sonicwall. This feature is poorly documented by Sonicwall, so I thought I'd pass it along. Has anyone else run into this, or figured out the rationale for it? I've used the iaxcomm softphone and a snom 200 behind serveral different sonicwalls over the past year or so without any problem. The sonicwall should not be a problem for iax calls at all. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users OK apart of my beleive that sonicwall is a piece of crap (personal), try to do a port forwarding for the IAX port (4569) regards saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with two cards Digium
Bart Fisher wrote: I'll guess the TE410P is being loaded first - Try swapping entries in zaptel.conf and zapata.conf Bart - Original Message - From: Bartosz Supczinski [EMAIL PROTECTED] To: Asterisk-Users asterisk-users@lists.digium.com; Asterisk-Dev asterisk-dev@lists.digium.com Sent: Tuesday, February 28, 2006 6:57 PM Subject: [Asterisk-Users] Problem with two cards Digium Hello, I`ve got a problem which I can`t deal with. I own 2 cards - TDM2400P and TE410P. I`ve put them into a HP Proliant DL380 G4 server, compiled the drivers according to the manual. Unfortunetly there are both cards channels are configured in zaptel.conf file the first module (in this file) sends an error. For configuration: span = 1, 1, 0, ccs, hdb3, crc4 fxsks = 21-24 bchan = 25-39, 41-55 dchan = 40 root# modprobe zaptel root# modprobe wctdm24xxp -- ZT_CHANCONFIG failed on channel 25: No such device or address (6) FATAL: Error running install command for wctdm24xxp For configuration: span = 1, 1, 0, ccs, hdb3, crc4 bchan = 1-15, 17-31 dchan = 16 fxsks = 145-148 root# modprobe zaptel root# modprobe wct4xxp -- ZT_CHANCONFIG failed on channel 145: No such device or address (6) FATAL: Error running install command for wct4xxp If the configuration applies only a single card the modules are loaded correctly. The analog card is equiped with one FXO module, which is placed in the last joint. I`ve disabled hyperthreading in my kernel and in BIOS, interrupts are not shared. Besides I`ve put the cards in other slots, switched on and off ACPI and other functions in BIOS as well as in my kernel. Maybe the problem is in the drivers? I`ve attached some information which might be useful. http://www.dir.pl/~supczinskib/logs.tgz -- With best regards Bartosz Supczinski IT Manager DIR Konstytucji 3 Maja 2 86-300 Grudziadz POLAND www.dir.pl t.: +48 (56) 6440100 f.: +48 (56) 6440111 m.: +48 (504) 019040 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Check dmesg toojust in case.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED COMMENT ON USING FEDORA CORE 3
Rich Adamson wrote: Can somebody share his experience with me in using fedora core 3 as asterisk server using quad port card (e1/pri) at full capacity. Runs fine and is very stable. Full capacity is 100% dependent on exactly what asterisk is doing (eg, transcoding), the PC hardware, etc, and has nothing to do with fc3. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-use Yes i totally agree... we took a Xeon 2.8 Ghz 533 FSB 1Gb Ram SuperServer and it took a little work to bring it to 100% without issues. if u can keep the load in the machine under 2 will work like a champ, what ever u use. regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
The VoIP Connection wrote: I have used every phone and talk to customers using different devices all day long and I can tell you there is no single IP phone that is perfect for everyone. You will not find the answer on a newsgroup or a wiki, you need to judge for yourself. For example, while I may love the decidedly euro ergonomics of the snom, you may find it impossibly unconventional. We have lots of customers who are very happy with their GXP-2000's as well as a number who are not. It depends on how they are being used (especially LAN or WAN) as well as the firmware version and networking environment. We also have many customers who love their Polycoms and there is no doubt that they build a quality product. They aren't cheap but they don't disappoint. By the way, Polycom officially supports Asterisk through certified resellers as of October 2005. Snoms are great also but they seem to be having some trouble getting the version 5.0 firmware stable. If you can live with the features in V4.x for a while, these phones are terrific. Probably the best overall integration with Asterisk of any IP phone currently available. Aastra seems to be getting it together at last and also are worthy of consideration. I sell phones for a living and here's what I recommend: First, select a reliable and competent vendor who will work with you (shameless plug for The VoIP Connection). Talk to them and narrow the field to a sampling of the phones you think will work for your organization. Set up a test scenario that simulates the network environment you will have and learn how to set the phones up with Asterisk (and vice-versa) so that they work the way they should. Learn how to use the features well enough to teach them (if you can't explain the basic operation of the phone in 5 minutes forget it), and then put them in front of a sampling of the people who will use them every day. Pay special attention to your receptionist and office manager since they will be the ones you will hear from the most. There really is no shortcut if you want your users to be happy. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: mustardman29 [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 21, 2006 12:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What business IP phone to use I have been struggling with this issue for about a year now. There were just too many IP phones to choose from at all sorts of price points and not enough information about any of them. Now I am looking at the situation again and if anything it has gotten worse. There are even more phones and all sorts of opinions. For every person that says phone x is great there is someone else complaining about it. I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I pretty much know what those two phones are about. Lot's of people talking about Polycom phones but they still seem to have their problems and since they don't officially support Asterisk I have my concerns. I really don't want to have to keep buying phones to find out for myself as it get's expensive real fast. Is there any unbiased comparison of various phones and features anywhere. If someone wrote a book I'd buy it but it would probably be obsolete before it was published with the rate of new IP phone introductions and firmware revisons. I hear some people praising the GXP2000 phones and I gotta wonder what they are smokin (regardless of firmware revison) so I just don't know who to believe anymore. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users From my point of view.. we tried grandstream 101/102 and the GXP 2000, we tried sipuras, polycoms and cisco.. and definitelly i put my bet for the polycoms.. now the GXP 2000 at his new prices probably will be a good answer, before at the same price that the polycoms don't have anything to do... budgetone don't ever bother u spend more time in support that his price. so at the long run u don't save anything. there are fine when u have 1 or 2.. but mass deployment :D that's another history... sipura 841 the only issue for me the spearker phone.. they are super stables but not to be used in a callcenter, they trend to brake.. i still think that an analog phone buy in what ever is better in callcenters that every other phone, but for have one in 1 office that don't need to use that much the speaker phone are super. cisco they are fine.. i still prefer polycoms. regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Test my test-branch!
Olle E Johansson wrote: Friends, The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback. For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is written by me. There are large contributions from other developers, code that I maintain in several open subversion branches in order to help them stay up to date with their work. To assist the testing group and make life easier, I've combined a lot of patches into one superbranch for testing. I've added the README further down. ** PLEASE help the community, please test this branch. Check it out like this svn checkout http://svn.digium.com/svn/asterisk/team/oej/test-this- branch test-trunk Then cd into test-trunk and run make then make install Report any bugs in the proper open bug in the bug tracker. If you like new functions, add a comment that this works for you. Provide feedback, make our work easier. Run svn update from time to time to get the latest version. Any changes from trunk will be merged into this code. Read the README.test-this-branch file to get more information. Thank you for your help! /Olle PS. Obviously, this is test code, not recommended to be closer than 2 miles from your production servers. - README.test-this-branch --- - TESTING BRANCH - WELCOME!! -- Asterisk is developed by the Asterisk.org user community. The development team does not only consist of coders, but also testers and people that write documentation and check for security problems. This is a combined branch of many patches and branches from the bug tracker that needs your testing. Please test and report your results in the bug tracker reports for each patch. What's in this branch? -- This branch includes the following branches: - sipdiversion: Additional support for the Diversion: header - jitterbuffer: Jitterbuffer for RTP in chan_sip (#3854) - videosupport: Improved support for video (#5427) - peermatch: New peer matching algorithm (no bug report yet) - rtcp: Improved support for RTCP (#2863) - dialplan-ami-events: Report dialplan reload in manager (#5741) - sipregister: A new registration architecture (#5834) - subscribemwi: Support for SIP subscription of MWI notification (#6390) Coming here soon: - iptos: New IPtos support, separate audio and signalling (#6355) - metermaids: Subscription support for parking lots (#5779) - multiparking: Multiple parking lots (#6113) And the following stand-alone patches - New CLI commands for global variables (#6506) - Additional options for the CHANNEL dialplan function All of these exist in the bug tracker * PEERMATCH: New object match for incoming calls. Skip the user :-) - In this code, we will match incoming calls like this: - First user on From: user name - Then peer on From: user name *** NEW - Then peer on IP and port This means that in most configurations, you can configure a phone entry as type=peer instead of type=friend. Subscriptions will work much better with just one object to match. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Olle.. i will very happy to test ur brand ... how i can download that. can u include the patch for silence support in ur brand? We have a lots of place that definitelly will have a good use for that. i can include 2 new events for ami based in CPU, memory and network utilization regard Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom ip601 attendant console
Damon Estep wrote: Anyone successfully set up one of the polycom soundpoint ip sidecars with asterisk to monitor and allow transfer to monitored extensions? How does it work? Any issues? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It works beautifull and not issues. regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2 in production w/100+ phones?
Christoph Eicke wrote: On Wednesday 18 January 2006 18:22, Peder @ NetworkOblivion wrote: Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime voicemail, etc), but I'm not sure if 1.2.1 is ready for primetime yet. Thanks. ... never touch a running system... I wouldn't upgrade if there wasn't any great new features that you cannot live without... Peder ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users There is something that said... Don't try to change what is not broken. anyway we have 1.2.1 with all the good stuff runing with 240 stations... a ride a little bumpy at the beginning now super relaxing. regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Save the Quintum before I throw it out a window....
Neil Bullock wrote: Well the subject line probably says it all. I have a Quintum D3000 which I'm supposed to be getting connected up to our Asterisk system. No matter what I try, neither username or authuser config works. I've also tried md5auth and it still refuses to register. Any one have a config they could share with me? Any help would be much appreciated. Neil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You probably don't will need registration to dial out.. check the wiki there is something about this there regards Saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP NOTIFY on REALTIME USERS/PEERS
Reto Kortas wrote: Hi! I've read in the asterisk docs (AstARA.html) that realtime users/peers can't be notified (MWI with SIP NOTIFY) when they have new voicemail messages, because their object are not persistent in memory?! But that's what we really need!!! Is their any work in progress to get these things working? Or are their any known workarounds? I'm using asterisk version 1.2.1 with OBDC access to an MySQL database. Thanks in advance Reto ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mine works.. i am using res_mysql without odbc and works perfect regards saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channel monitoring whisper mode?
Script Head wrote: As this isn't a part of *, has anyone accompilished a whisper mode in yet? What I am looking for is an ability for to say something while monitoring a channel and the agent being able to hear what I say while the called party is not. ScriptHead ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I am thinking to develop one. regards saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budge Tone 102
Tomislav Parcina wrote: I have Grandstream Budge Tone 102 with Software Version:Program-- 1.0.5.18Bootloader-- 1.0.0.21HTML-- 1.0.0.42VOC-- 1.0.0.7. I'm planning to upgrade it with Firmware 1.0.6.7. My question is, does anybody has any ishues with this firmware version? Should I put this or some older firmware? I have 1.0.67 and for some weird reason the phones froze ... not sure yet why. regards saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_oss.so
Tomislav Parcina wrote: What does this channel do? Today I installed * 1.2.1 for the first time and I needed to put noload = chan_oss.so in modules section of modules.conf file. Will I miss some Asterisk functionality now? Just dial from the console check that your Dial cmd disappear from ur CLI. regards saul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma Asterisk at home
Jess Coburn wrote: Guys, I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recompile the zaptel but I can't seem to find the source, etc on the server after asteriskathome installs. Jess ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I use the regular asterisk at home.. with a tdm card and a sangoma card... we have an small home business and we use a manager we develop for it.. excelents results www.cripiland.com/screenshots/manager3.jpg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large Implementation
Dario M. Colombo wrote: Hi, somebody has implemented Asterisk in one organizacion with amount of extenciones in the order of 20.000? Thanks. 2 in 1 building? WOW that's a HELL of a PBX the maximum that we are achive in 1 BUILDING is 1000 extension regards Saul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I know if I have CRC-CCITT (README.Linux26)
Chuck Bunn wrote: You have lsmod | egrep crc_ccitt check in your kernel modules looking for crc_ccitt but FC4 comes with that regards Saul Hi, I have downloaded the latest release candidate v1-2-0-rc2 and I was checking the readme's in the zaptel directory and I came across a requirement I have not seen before. In the readme it says that for Linux 2.6 kernel you will need to have the 'CRC-CITT' functions compiled with the kernel. I am using Fedora 4 kernel version 2.6.13-1.1532.FC4. How can I verify that I have these functions? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline
Julio Arruda wrote: I was testing Broadvoice few weeks before Hurricane Wilma here in FL. Since then, I had been since the landline (Bellsouth), and I had to 'remote callfwd' the BS # to my broadvoice #. So, from my impression, is ok for my needs (I got a weird no ringback problem that I kind of solved with a Background trick), and no surprises yet regarding the bill (my mother in law call Brazil a lot from my house, no, she is not aware of the 'unlimited' plan. So I may be in for a surprise in a couple of months). I've no tried several calls at the same time, you may want to ask them.. PS: I'm running Asterisk 1.0.9 Dane Reugger wrote: We are considering Quantumvoice as a provider - They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 + extras - this seems considerable less expensive than the competition which seem to focus on. My second choice is BroadVoice $29.99 + $9.99 per additional line (in state only?) - more expensive, less features, and they don't seem loved by many ? Is anyone else using Quantum Voice? It was mentioned earlier that it requires an ATA connection and Asterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it? Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several calls at a time? Any advice or recommendations appreciated - I want to port my number but I'm running out of time and must make a decision very soon. Thanks, Dane Reugger Crescent City Technologies New Orleans, LA 70112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Broadvoice only allows only the normal 3 way calling so is 2 channels for # about BV i got a lot of water under the bridge every works ok supper ok for times. then BV brokes without you make a single change in your asterisk server and stop working.. if u call support you are the guy with the problem.. yes BV support sucks, and it took me 9 phone calls, 12 emails, 3 chargeback and 2 call to my bank to remove myself from their billing all them well documented... so my advice nothing can be worts than BV. regards Saul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sill looking for a provider
[EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote on 11/04/2005 04:34:18 PM: Try calleveryone.com Yes.. I have blown their trumpet before. They are a very good company with great support. Do they support IAX or just SIP? I've been reluctant to use a SIP provider for a number of reasons, including difficulties in using it through a NAT firewall or the fact that I have to open 10,000 ports on my Asterisk server! Am I overly paranoid? Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I spend three months trying to get me out of broadvoice... i hope they are better now.. good luck regards saul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue scheduling...
Scott wrote: Is it possible to schedule dymanic queues? Currently I have a queue that has dynamic members of which I would like to set a schedule for. From say 8am to 5pm the queue would ring the phones of queue members but after 5pm the caller get's VM. Is this possible? Thanks. Scott. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users A simple ivr in the dialplan.. i think will solve this easilly ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue scheduling...
Scott wrote: Is it possible to schedule dymanic queues? Currently I have a queue that has dynamic members of which I would like to set a schedule for. From say 8am to 5pm the queue would ring the phones of queue members but after 5pm the caller get's VM. Is this possible? Thanks. Scott. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users A simple ivr in the dialplan.. i think will solve this easilly check gotoiftime regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Webui to show registered phones
Hi For those who are insterested in monitoring and managing easilly the asterisk server.. this is a solution for multitenant hosted PBX o single tenant is windows based (the admin of couse) and http://www.cripiland.com/screenshots/manager3.jpg http://www.cripiland.com/screenshots/manager4.jpg http://www.cripiland.com/screenshots/manager1.jpg http://www.cripiland.com/screenshots/manager2.jpg regards Saul Matt Gibson wrote: Hi Guys, Here's what I use to view the current IAX and SIP peer status. It's not very pretty, but it works. I also have an included script (vm.php) that will show the current voicemail usage for a box. Uses php asterisk library to work through asterisk manager. Configure your options in cfg.php Matt Nicolás Gudiño wrote: Hi all, does anyone know if there is any app/webui that can show phones that are currently registered to *. I guess this sort of funcionality counld be grabbed from the CLI with iax2 show peers and sip show peers, but having little programming knowledge wouldn't know where to start. I'm asking because we currently have several sip phones onsite and lots of remote iax2 users who would like to see availability without dialing. plugYou can try with the Flash Operator Panel/plug http://www.asternic.org , it does all sort of things including sip and iax availability (you have to enable qualify for them). Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Join Event
Tressler, Joshua A wrote: I did a quick Google search of the lists and I hope that I am not asking a question that has already been answered recently. I have been working on a interface to use with our CRM software. I am using the manager interface and mysql to store the changes. The only issue I am having is when a caller joins the queue. Currently, I can show the status of phones (ready, not ready, ringing, ringing ack, in call, etc). What I am wanting to do is to be able to track the status of the call in the database and do things with it accordingly. I am able to accomplish this and make it work exactly as I want, but it requires a modification to the source. For some reason, the JOIN event in the manager interface doesn’t seem to have the unique call id. Almost every other event does, but JOIN doesn’t for some reason. Can anyone explain why it doesn’t? My boss asked us to remove our hack to the source and find another way as it we want to be able to update versions of asterisk and not modify the source. I thought that I could get around this by using the NEWEXTEN event that happens just before the join, but I can’t tie the two events together. I think you boss was right.. that will allow 100% compatibility with all asterisk versions as far the events are there lets see the events for a moment HMMM the call enter go to the ivr so u will get 1 event like this for the entire IVR Event: Newexten Privilege: call,all Channel: SIP/s-f36c Context: open Extension: 0606 Priority: 7 Application: Queue AppData: operator Uniqueid: 1127422073.9183 Server: asterisk1 Ok this is the JOIN event still u can find things there that will allow u to relate the call see the channel parameter. Event: Join Privilege: call,all Channel: SIP/s-f36c CallerID: ... CallerIDName: ... Queue: operator Position: 1 Count: 1 Server: asterisk1 Now the asterisk is preparing himself for call an agent... Event: Newchannel Privilege: call,all Channel: SIP/9915004-e198 State: Down CallerID: unknown CallerIDName: unknown Uniqueid: 1127422096.9194 Server: asterisk1 Event: QueueMemberStatus Privilege: agent,all Queue: operator Location: SIP/9915004 Membership: static Penalty: 0 CallsTaken: 86 LastCall: 1127422074 Status: 0 Paused: 0 Server: asterisk1 Ahh magic again a wait to relate see the channelcalling and i bet what ever u want that when the agent finish u will find the same similarities. Event: AgentCalled Privilege: agent,all AgentCalled: SIP/9915004 ChannelCalling: SIP/s-f36c CallerID: . CallerIDName: ... Context: open Extension: 0606 Priority: 7 Server: asterisk1 so following the events for channel u will probably able to do the same even without the uniqueid. 2 concurrent calls will have diferent channels always u just have to be carefull to ensure u follow the call from the beginning to the end. regards saul Basically, with the hack modified, here’s what I do: Call comes in, enter the info into the database with uniqueid as the key. When a call is answered, I update that record in the database and so on. Without the uniqueid on the JOIN event I am stuck. Any suggestions on a way around this, or a better way of doing it? I would also be curious if anyone would share their setup if the are attempting the same. Thanks, Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Join Event
Tressler, Joshua A wrote: Saul, What you are suggesting follows along the lines of what I am currently trying however I have determined that if the incoming call has no callerid, then the channel name is just Zap/1-1/ . For some reason asterisk doesn't even add the - to the end of the channel name In zap channels :) zap/1-1 is the - u don't will get zap/1-2 at least i have a 300 phones system runing and never got 1... b/c only will happen when u send another call in the same channel... if the user hangs up :) u will get another event.. i think i can look here for that event too if u need .. and the same will happen if the agent hangup u get another event etc etc Event: Leave Privilege: call,all Channel: SIP/s-f36c Queue: operator Count: 0 Server: asterisk1 Database problems can be reduced to the min.. u can creat an id unique for u when u move to the db but in the asterisk enviroment the only way u have to follow the call is the channel...and if the channel is renamed b/c u got a park call or something u will get the rename event regard Saul My concern is that we could get a call that wouldn't go completely through the queue (aka, the user hangs up, or a db problem) and then an hour later get another private call on the same Zap/1-1/ channel and then we could have an issue of the uniqueness of the record. Do you follow my scenario? I really thought that this may work until the above problem. Do you have any to this issue? Again, I appreciate your help with this. Thanks again, Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source Billing Software
:) trabas and asterisk have big misunderstanding the don't thing to work like it should be :) regards saul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Application Development in Asterisk
Alex Vishnev wrote:___ Write an asterisk application will be better ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a dialup connection?
Hi We try sucessfully a firefly phone with g729 in a very low bandwidth scenario... regards saul diaz Cripiland LC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
Hi We have 11 DID with broadvoices... So far so good regards Saul Diaz Cripiland LC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP - Billing Solutions with Asterisk?
Roman Zhovtulya wrote: Hello We was in the same situation about the same time. we take the a version of rate_engine (routecall APP) and modify the implementation now our solutions is able to find a least cost route and at the same time show all the cost and times even with prepaid solution or postpaid solution. Same about the look in our prototype. regards Saul Hello, I was in the same situation about half a year ago when I evaluated the billing systems for our Asterisk setup. Sice I was already putting the cdrs in MySQL and had all the users and extensions there as well, my solution was to develop our own rating engine (jsp - server-side-java- based) and integrate it into our existing portal solution, where we keep all the users, etc. It works really good and allows a real-time view on all the costs, duration, etc of the calls made. All the statistic is neatly displayed in the web-front-end where every user can see all their information. If you are interested in taking a look at our solution, just mail me at [EMAIL PROTECTED] and Ill get you a test account. Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Dienstag, 22. Mrz 2005 17:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] VOIP - Billing Solutions with Asterisk? Hi, With everyone other that who uses Asterisk.. what is the best solution you have found for billing VoIP users? Radius? Just parsing CDR reports nightly? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP - Billing Solutions with Asterisk?
Roman Zhovtulya wrote: What the web interface running on (JSP, PHP, ASP, etc)? PHP. based in templates, RouteCall deal with all based in cdr, and the system deals with the rest you just need asterisk realtime (SORRY CVS HEAD) I think that can be improved but u can have different rates for handling systems, can handled even some asterisk functions like extensions, peers. etc. and the billing. CDR information Regards Saul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Saul Diaz Sent: Dienstag, 22. Mrz 2005 20:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VOIP - Billing Solutions with Asterisk? Roman Zhovtulya wrote: Hello We was in the same situation about the same time. we take the a version of rate_engine (routecall APP) and modify the implementation now our solutions is able to find a least cost route and at the same time show all the cost and times even with prepaid solution or postpaid solution. Same about the look in our prototype. regards Saul Hello, I was in the same situation about half a year ago when I evaluated the billing systems for our Asterisk setup. Sice I was already putting the cdrs in MySQL and had all the users and extensions there as well, my solution was to develop our own rating engine (jsp - server-side-java- based) and integrate it into our existing portal solution, where we keep all the users, etc. It works really good and allows a real-time view on all the costs, duration, etc of the calls made. All the statistic is neatly displayed in the web-front-end where every user can see all their information. If you are interested in taking a look at our solution, just ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
