Re: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
A T carrier cable is not the same as an ethernet cable. A T carrier cable uses a real metal shielded RJ-45 and loosely twisted pair wire. With most modern T carrier equipment, you can use a CAT-5 ethernet cable instead of a real T carrier cable. A T-carrier crossover cable does not have the same wiring pattern as a crossover ethernet cable. With an older piece of equipment like the Matra, I would be tempted to purchase a real T carrier crossover cable. This is covered in my book, by the way. Louis-David Mitterrand wrote: Hello, I am about to put an asterisk server between the telco E1 and our old Matra PBX. Should I use an ethernet cross cable? Something else? Paul [EMAIL PROTECTED] www.signate.com ___ --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: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] whatcable to connect a legacy PBX to a TE410P ?
I agree. I haven't had a problem using CAT-5, even for long runs, however it's not a real T-Carrier cable and I didn't know how old the PBX is. Paul I have not in my experience seen any problems with using a Good Quality Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you should be fine. As far as the shielding goes, I use UTP cables and Connectors all the time and some of my X-connects run over 100 feet. Paul Mahler - [EMAIL PROTECTED] www.signate.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Signate Intro to * - London Training March 21-23
We still have a seat open in the London Introduction to Asterisk class. TKS Paul Paul Mahler [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of McQuiggan, Mark xt46480 Sent: Sunday, March 05, 2006 12:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with libpri? While testing a problem with spontaeously and occasionally rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data="" at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function pri_release_timeout, and line 2589 reads: if (pri-debug PRI_DEBUG_Q931_STATE) pri_message(pri, Timed out looking for release complete\n); PRI Debug was not on in the asterisk console. Any ideas? My asterisk restarts about twice a day, and drops any current calls in the process. Regards, Mark McQuiggan This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intro to Asterisk VoIP telephony course - March 21st London seats still available
There are still seats open in our March 21st to 23rd Introduction to Asterisk and VoIP telephony course. More information is available at www.signate.com. Paul Mahler ___ --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] 5,000 concurrent calls system rollout question
Based on our benchmarking, I am VERY skeptical of this number. Im guessing that you dont really have RTP streams going through the NIC. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joash Herbrink Sent: Wednesday, February 01, 2006 12:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout question I have tested an asterisk server with over 5000 concurrent calls. The system setup was a P4 HT 3Ghz, 4 Gb RAM, and 1 gbps Ethernet connection on a cisco 3560 switch. This works, but puts some serious stresses on the system. Why don't u considered using g.729 codec, this will at least lower the bandwidth consumption significantly, and, you can overcome the CPU resource issue by just using a server grade multi CPU xeon server. I would never the less still connect the system via 2 ethernet connections, just for some redundancy, as mentioned before in this thread. Bandwidth should be about 24 kbps (half duplex) per call So, 5000 * 24 is roughly 120 mbps, so a gigabit Ethernet should do just fine. Joash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Wildes Sent: Wednesday, February 01, 2006 8:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Dinesh Nair wrote: On 02/01/06 09:29 Damon Estep said the following: Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps - full duplex. Have you ever seen a NIC or switch that can run GigE full duplex at 80% utilization and not at least start to fall apart? additionally, 5000 simultaneous SIP calls at 20ms intervals will send, 5,000 * 50 * 2 = 500,000 packets per second (full duplex). not too many boxes can handle such packet load, in spite of the relatively small packet sizes. Why not bond multiple NICs together to do a load balance output? Would provide redundancy as well. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Have you verified that you are actually sending sound over the RTP streams? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith Sent: Friday, January 27, 2006 11:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question What's you mix of calls going SIP/IAXand to PSTN? We've done some benchmark experiments on a 3GHz HT box with 1GB of ram, mirrored traditional IDE disks. The box has a Digium quad-PRI, a TDM40B, a TDM22B and a Sirrix quad-BRI board in it. This box can run 120 active calls over 4 PRI spans. Its running MusicOnHold into 60 of the channels, playing various GSM prompts into the other 60. The user cpu usage is about 25%, the system cpu about 25% also. We can add to that 5000 registered SIP peers and 5000 registered IAX2 peers - total of about 100 registration refreshes per second. That adds about 40% more user CPU and pretty much fills up CPU. Audio quality is still perfectly fine, and PRI slips few and far between. Load average for the whole mix sits between 5 and 10. About 550Kb/s out and 400Kb/s out on the ethernet for the registration traffic. Also on www.voip-info.org - search for dimensioning Rob On 1/28/06, Vic [EMAIL PROTECTED] wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 5,000 concurrent calls system rollout question
Signate sells a single server that can get you to the call volumes you need. Paul Mahler [EMAIL PROTECTED] www.signate.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Saturday, January 28, 2006 7:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, canwe make ita bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Still an open Seat in London for Next Weeks Signate intro to Asterisk Course
We still have a seat open in our Asterisk training course next week in London. You can find more information at our Web site, www.signate.com I'm going to be teaching the class. Paul ___ --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] Second edition of my * book has been released
Hi Greg, My book is a good place for a beginner to get started. I also find it to be useful as a reference for Asterisk. It's not an advanced book, there are advanced features it doesn't cover, for example AGI or the management interface. It should be very helpful for your customers. It should be helpful for a beginning to intermediate administrator. I still frequently refer to it myself when I'm having a senior moment. :) There isn't anything in the book that would make it less useful for the CVS or stable branches. The O'Reilly book is excellent. I think my book complements the O'Reilly book. If I were just starting I would buy both. I think my book may be a bit more useful as a reference. I think I cover a bit more beginner's territory. Hope This Helps, Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, January 09, 2006 9:10 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Second edition of my * book has been released How does it compare with the O'Rielly book? Does it include information on CVS, or primarily on stable? Can it be provided to customers, or is it more sysadmin oriented? Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Thursday, January 05, 2006 9:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Second edition of my * book has been released The second edition of my Asterisk book VoIP Telephony with Asterisk is now in print. It's reorganized and expanded. TKS Paul Mahler Paul Mahler [EMAIL PROTECTED] www.signate.com ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006 ___ --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] Recommendations on a WiFi phone for *?
I would be MUCH more tempted to use an IAXy or SIP adaptor and a cordless phone. It will be less expensive and it will likely work better. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joash Herbrink Sent: Monday, January 09, 2006 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *? The zyxel p2000W Works fine, good batt. Live. Decent sound quality. All in all a good product for about 150 euro's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Edelbrock Sent: Tuesday, January 10, 2006 2:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Recommendations on a WiFi phone for *? We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The second edition of my Asterisk book is now available
The second edition of my book VoIP Telephony with Asterisk is now in print and available. You can find out more about it at our web site http://www.signate.com/products.php This book is written for beginners. It will make it easier for you to get started. The second edition is reorganized and expanded. Thanks, Paul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Second edition of my * book has been released
The second edition of my Asterisk book VoIP Telephony with Asterisk is now in print. It's reorganized and expanded. TKS Paul Mahler Paul Mahler [EMAIL PROTECTED] www.signate.com ___ --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] Is there a GUI for asterisk realtime
We sell a complete Web facing interface called sigMan that works with realtime. http://www.signate.com Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Friday, December 23, 2005 10:40 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is there a GUI for asterisk realtime There is a web interface. It's pretty basic but you can find a demo here: http://dc.maxnet.ru/cpdemo/ I know the guy that owns it. Contact me if you're interested. It's payware. Darren Wiebe [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, Is there a GUI to manage the users in database (realtime) ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.5/212 - Release Date: 12/23/2005 ___ --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] Avaya 4612 IP phones with Asterisk?
Thank you very much for trying it for me, Dave. I really appreciate it. Paul _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Rahn Sent: Tuesday, November 08, 2005 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Avaya 4612 IP phones with Asterisk? I have gotten 4620's to work ( convert to sip ) It works ok... at best. I have a 4612 at work I will try tomorrow. good luck with yours . Dave _ From: [EMAIL PROTECTED] on behalf of Paul Mahler Sent: Tue 11/8/2005 2:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Avaya 4612 IP phones with Asterisk? Has anyone been able to make these phones work with *? If you have, what does it take? Thanks! Paul Paul Mahler [EMAIL PROTECTED] www.signate.com ___ --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 attachment: winmail.dat___ --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
[Asterisk-Users] Avaya 4612 IP phones with Asterisk?
Has anyone been able to make these phones work with *? If you have, what does it take? Thanks! Paul Paul Mahler [EMAIL PROTECTED] www.signate.com ___ --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] Asterisk as an internal pbs for a samall company
Why do you want to use a SIP provider instead of a PSTN connection? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Taylor Sent: Wednesday, November 02, 2005 4:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] Asterisk as an internal pbs for a samall company Well, U right, many missing informations. The case is quite simple(I guess), we have dids, and each call to these dids has to be routed to the right handset thru Asterisk, no Ivr at this time, at least an answering machine in case of busy or not available users. For the rest, we need to be able to have external calls to pstn, or even to other sip phones form other providers. Is that enough? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de trixter aka Bret McDanel Envoyé : mercredi 2 novembre 2005 13:48 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk as an internal pbs for a samall company On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote: Hello all, We'd like to use asteriek as an internal pbx connected to an external sip provider to make outbound/inbound calls to pstn. We have the provider and have installed an asterisk at the office. Does anyone have a sample config? We need 25 telephone numbers(dids), to be registerd to the provider and be able to ceceive calls. Any advice is welcome. Sorry for the noob question, Olivier What you want to do depends largely on what you want to do. While that seems like a cylic statement I will try to explain. You have said that you want to route calls between your asterisk box and the PSTN via a VoIP provider that you have. So far that seems simple, but how are those calls going to go bewteen the office workers and asterisk? You will need configurations for that. How are the inbound calls going to be routed? Via an IVR? Well you will have to configure that. There is a lot of information that is missing from this setup. www.voip-info.org has a lot of asterisk examples including configuration files. You may find something there that does what you want. I cant easily help you solve this problem (and suspect that no one else can either) until you provide more information on exactly what you want. If you wish to discuss this offl ist feel free to email me directly. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/155 - Release Date: 11/1/2005 ___ --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] voip provider in a box
We have a turn-key solution available that does exactly what you are asking for. You can reach someone for more information at 415.442.4010. TKS Paul [EMAIL PROTECTED] trixter aka Bret McDanel wrote: I am tasked with evaluating ready made solutions for a voip provider. Does anyone have any recommendations for software, specifically the environment will be a chargable voip provider (ie broadvoice, vonage, etc). They wanted me to see what was there and write something if nothing they like exists. 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
Re: [Asterisk-Users] Soekris and Asterisk
You need about 30MHz per channel. That means the Soekris can only handle part of a T1, it will never handle a quad span. Paul --- Kristian Kielhofner [EMAIL PROTECTED] wrote: Craig Guy wrote: Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig Craig, It all depends on where you are going to do what (PRI, echo cancel, etc). Also, for four spans the interrupt load alone could probably saturate the CPU. If you want to try, AstLinux will be an excellent start... http://www.astlinux.org P.S. - I created AstLinux, so of course I would recommend it! -- Kristian Kielhofner ___ --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 Paul Mahler [EMAIL PROTECTED] www.signate.com ___ --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] Dual PRI fail over
If you have a PRI, many vendors will support sending calls to an alternate destination if the T1 is down. SBC, for example, calls this enhanced alternte routing. If the T1 fails, call are routed to the destination of your choice at the SS7 switch. Paul [EMAIL PROTECTED] --- Tom [EMAIL PROTECTED] wrote: I currently have a single PRI however we are getting a second PRI, and the provider (qwest) wants to know if our PBX supports GSAS (they say its a redundant d-channel technology but searching on google for GSAS reveals less than nothing). I've set something similar up before on a cisco 5350, where if one of the PRIs fails, all of the calls destined for either PRI will be routed down the one that didn't fail. Basically the 2 PRIs are bonded together, and act as one. During normal operation the calls come down each PRI in a load balanced fashion (IE if I've got 30 calls up, 15 will be on one PRI and 15 on the other). Is there any way to set something similar to this up in Asterisk? Tom ___ --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 Paul Mahler [EMAIL PROTECTED] www.signate.com ___ --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
[Asterisk-Users] Help - Cisco router configuration with Asterisk
I am looking for someone who knows how to configure cisco routers to work with *. You can contact me at Paul Mahler [EMAIL PROTECTED] Thanks! Paul Paul Mahler www.signate.com ___ --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
[Asterisk-Users] SIP Benchmarking / Stress Testing
we used sipp, the opeh source benchmarking software sponsored by HP. We can send you our benchmark, if you like. We did run into a problem, though. The benchmark suite core dumps on us at about 5100 simultaneous SIP streams. Regards, Paul Paul Mahler www.signate.com ___ --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
[Asterisk-Users] IAX2 Softphone Quality Network Cards
I have had uniformly bad experiences with soft phones when there are network issues. Hardware phones seem to work much better if there are network problems. For example, I have been able to make fine calls over a wireless link I use with a cisco 7960, but NO softphone works over the same link. You should also look at the settings for the NIC on the computer. Are your network equipment and NIC both set for full or half duplex? They should be set to the same duplex setting. GIG equipment is easier, it defaults to full duplex. Hope this helps! Paul On 8/26/05, Adam Robins [EMAIL PROTECTED] wrote: We are in the process of an Asterisk call center deployment using IAX2 G711 ulaw softphones. Outbound sound quality is terrible. This week we rebuilt the entire LAN with Cisco 2950-EI switches and have employed QoS on the switches and router. Still sounds terrible. What we are now finding is that the network card in the PC may be the key to the problem. A Dell Optiplex P4 2.4GHz 512MB machine with an onboard Intel NIC is bad, while an older Dell Dimension P3 864MHz 128MB machine with onboard 3COM sounds good. Has anyone out there had a similar experience? Thanks, Adam Paul Mahler www.signate.com ___ --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
[Asterisk-Users] [Asterisk-Dev] Job Opening - Release Engineer
Signate has an immediate opening for a qa/release engineer for our line of VoIP telephony products. Release Engineer Signate is rapidly growing and profitable. We are about to launch a new line of telephone software products. Thats where you can come into the picture. You would support Signate's software development team by reviewing new and changed code, tracking and auditing change histories, debugging build and runtime problems, and maintaining a build process to support ongoing RD and regression and user/system level tests. As a Release Engineer you will have primary responsibility for updating release branches in our source control system, building and testing release binaries, and pushing releases to production. You will design and document improvements to the integration / build / test and release processes. Our development team is distributed around the world, and you could be located anywhere. If you have a passion for testing, are a quick learner, self-motivated and capable of working independently as an integral part of a team wed like to talk to you!. Job Requirements: Minimum of three years' software QA and configuration management experience. Genuine enjoyment of SQA work. Strong knowledge of Internet technologies, mySQL, PHP and the Linux operating system. Exposure to XML/XSL and JSP. Experience with c and Asterisk source code. Proficiency with software testing automation tools. Ability to create effective test plans. Ability to prioritize problems in problem tracking software applications Experience with software configuration management systems / source code version control systems. Must have excellent technical writing and communication skills, and strong problem solving skills. Send your resume and salary requirements to Paul Mahler at [EMAIL PROTECTED] Paul Mahler www.signate.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wildcard/FXO config
Note that the single line card will not work in a variety of more recent servers including Dell servers. First, you have to get the card configured to run with Linux. This means loading the correct driver and then configuring the driver. The driver configuration information is held in the file /etc/zaptel.conf. Here is what you need in zaptel.conf to configure your single port FXO board: loadzone=us defaultzone=us fxsls=1 ; fxl interface on fxo port ; channel one ; loop start The command ztcfg uses the contents of the file, zaptel.conf, to configure the driver. The file /etc/asterisk/zapata.conf contains the information used to configure * for the board. Here is a working example of a configuration for the single line FXO card. ; zapta.conf configuration file ; Contact : [EMAIL PROTECTED] [channels] language=en context=main signalling=fxs_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 immediate=no context=main callerid=Your Name 555-1212 channel = 1 Hope this helps. Paul Mahler [EMAIL PROTECTED] http://www.signate.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
[Asterisk-Users] Portable USB headset for VoIP
I've bought bunches of these: http://www.tigernetcom.com/products_USB_100.html they work great. Very handy. Paul [EMAIL PROTECTED] I'm trying to find a voip-suitable USB headset (I.E. headphones + microphone) which I can use with my laptop while I'm traveling and using Firefly or another softphone. I'm currently using a Logitech headset which works well (except the echo it generates toward the other caller when I turn up the gains too high), but it just doesn't carry well - in fact, I can't carry it in my laptop case any more just becuase it doesn't fit and it was getting very beat up. I'd like to find something which folds up and is designed for travel. It has to be USB sicne I don't have a MIC in (just line) on my laptop. Any ideas? -forrest ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 267.4.1 - Release Date: 6/2/2005 Paul Mahler www.signate.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
[Asterisk-Users] Astersik vs. Pingtel
Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.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
[Asterisk-Users] FREE music for downloading
Need new Music on Hold for your PBX? Signate is happy to make a variety of classical music selections available, sampled at rates that are appropriate for telephony. There is no charge. The selections feature Elena Kuschnerova, pianist, and Lev Guelbard, violinist, playing public domain pieces that will give callers a classic impression of you or your company . Click on the link to see a list of the available music and download what you want from our ftp site. http://www.signate.com/moh.php Thanks to Greg Camp, who graciously provided us with the original files. We plan to add other types of music over time. Legal Stuff Follows SIGNATE MAKES NO WARRANTIES, EXPRESS OR IMPLIED, REGARDING THE FREE MUSIC ON HOLD FILES, INCLUDING, WITHOUT LIMITATION, ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. SIGNATE SHALL NOT BE LIABLE TO YOU OR ANY OTHER PERSON OR ENTITY FOR ANY GENERAL, PUNITIVE, SPECIAL, DIRECT, INDIRECT, CONSEQUENTIAL OR INCIDENTAL DAMAGES, OR LOST PROFITS OR ANY OTHER DAMAGES, COSTS OR LOSSES ARISING OUT OF YOUR USE OF THE FREE MUSIC ON HOLD FILES, EVEN IF SIGNATE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES, COSTS OR LOSSES. Paul Mahler www.signate.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] OT: USB handsets / softphones
I agree, this is a fun device. It's a lot easier to use than a headset. The sound quality is excellent. Just don't turn up the volume too much or you will get a lot of echo. Echo is less of a problem with a good usb headset. It's a little quirky. All the sound from your pc gets routed to the phone. You can set x-lite to send ringing elsewhere. You have to load the driver that comes with the phone to be able to dial from the phone keypad. to dial a call, you press the dail button, dial the number, and press the dial button again. I can stand by the USB U2 Phone sold at http://www.eezeephone.com connected to a Firefly Third Party Version of the Softphone. This is one of the best combos I have ever used. Voice quality is phenomenal when using GSM or ILBC at just one end (better if on bothe ends) Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Friday, April 15, 2005 2:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT: USB handsets / softphones Here is just my personal opinion on the whole thing as I spent a good deal of time on this myself. In the end I had MUCH better results, and better sound quality moving to a Sipura SPA-1001 and a $14.99 cordless phone (with $12 rebate at Best Buy). Not only does it sound better, I don't have to walk around carrying my huge laptop. Full review of the SPA-1001 will be on GeekGazette tonight. Kerry http://geekgazette.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: Friday, April 15, 2005 11:09 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] OT: USB handsets / softphones Hi all, After googling around and searching both * and xten archives, I was still unable to find a working pair of softphone/usb *handset* that work with both keypad operating the softphones buttons *and* working incoming call ringer on the handset. I'm hoping that, while being OT for * discussion, someone else on this list had luck with finding a pair that works, preferably with xten's xlite/xpro. Any feedback is appreciated. regards, Vahan ___ 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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.13 - Release Date: 4/16/2005 Paul Mahler www.signate.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
[Asterisk-Users] cisco 7960 SIP setup
There's a long chapter in my book about re-programming the 7960 from skinny to SIP that might help you out. Figuring it out was non-trivial. You can get the book at Amazon. TKS, Paul Mahler I can't get the 7960 to reconfigure and work. I am a newbie to voip. I went through the list and read some other comments about the 7960 and unlocking it. It is a used 7960 that came with CallManager. I need to have SIP. I first reset the phone to factory defaults then I changed the TFTP server address in the settings. I have unlocked the phone with **# and it shows the lock as unlocked in the upper right hand corner. I was told that the phone should be able to download the SIP... file once the TFTP address was changed. So far nothing though. Any ideas? Mike Paul Mahler www.signate.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
[Asterisk-Users] Many analog lines
You can purchase a T1 card and a channel bank. Or you can buy a gateway device, for example http://www.mediatrix.com/products_devices.php?prodid=3 We have been having good luck with the Mediatrix gateways. be very careful with your choice of vendors for a gateway. I had what I thought was the worst support of my entire career with Audiocodes. I found the documentation to be useless, too. My advice is to stay far, far away from Audiocodes. Hi, how to use Asterisk where I need to have lets say 40 analog lines. Any ideas? Thanks, David ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.6 - Release Date: 3/30/2005 Paul Mahler www.signate.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
[Asterisk-Users] We just released our new Asterisk Installation CD set. with 24/7 monitoring
Here's our recent announcement of our new Asterisk Installation CD set: Signate has announced its new Asterisk Installation 2005 CD Set. It's, a complete software PBX (private branch exchange) telephony appliance in a single package. The CD set installs Linux pre-configured for telephony, a stable 1.0x distribution of the open source Asterisk PBX, and Signate's optional, free PBX monitoring. When Signate's Asterisk Installation 2005 CD set is loaded onto a PC with an internet or PSTN telephone connection, it creates a running VoIP PBX ready for configuration in about twenty minutes. SigMON, Signate's included PBX monitoring software, helps keep the PBX running. SigMON monitors about 20 different conditions on the PBX and sends alerts if a condition needs to be attended to. Monitored conditions range from hardware conditions such as available disk space and CPU utilization, software conditions such as whether the PBX is running, and telephony conditions such the state of connections to telecommunications providers. One instance of Signates PBX monitoring service is free for the PBX created by a Signate Asterisk Installation 2005 CD set. Signates VoIP Telephony with Asterisk Book and CD Set is $89.95 and Signate's Asterisk Installation 2005 CD is $49.95. They are available at Amazon, Signate or Ebay. Paul Mahler www.signate.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
[Asterisk-Users] Forklift a 2000 phone PBX
We sell an Asterisk based soft switch that starts at 5000 simultaneous connections and goes up from there. paul [EMAIL PROTECTED] I'm staring at an RFP--this company wants to replace a 2000 position PBX (at eight locations) with a new system. Their mindset is Nortel/Avaya because they talk about 28-button digital sets. The do specify a few IP phones for just one location, so they are aware of VoIP. I'm going to bid on this--there's nothing to lose except the time it takes to write the proposal. I'll bid an off site Asterisk system with SIP telephones. Using the metric of 100 SIP phones/box, I'll bid twenty Asterisk boxes with ten boxes at each of two hosting locations. Each phone will have registrations to both sites. The big unknown is wiring. I'm going to assume the worst, that the existing LAN is overloaded. I would a) have to make LAN wiring out of existing Cat3 wiring, or b) install a new voice-only LAN. Does anyone know how to qualify existing Cat3 wiring for use as a LAN? Has anyone does an Asterisk system on this scale? Thanks for your help, Mike P.S. Sorry for the cross post, but I would like everyone to see this. ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.1 - Release Date: 3/23/2005 Paul Mahler www.signate.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
[Asterisk-Users] Digium T1 Card Questions
We use a lot of Digium single span T1 cards. They work great. They operate just fine on a T1 without ISDN. Digium will support the card for you. Paul [EMAIL PROTECTED] I have a couple of questions about Digium's T1 cards, such as the TE410P. Any answers would be greatly appreciated. 1) Do they support standard T1s or are they ISDN-only? 2) Do you know of anyone offering support for configuring T1s for Digium cards, and if so at what cost? Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.1 - Release Date: 3/23/2005 Paul Mahler www.signate.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] PRI Question
You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work. You have to install the Linux packages to split the line. NON trival. Works great, though. Paul Paul Mahler [EMAIL PROTECTED] www.signate.com On Mon, 2005-03-21 at 21:16 -0700, Tim Chandler wrote: Let me further clarify this. I am looking to buy the TE110P. The website says that The TE110P with Asterisk will route voice and data traffic, and eliminate the need for an external router. How does this work? How is the data transferred - as a pass-through like a NAT to the server's network card? What kind of network slowdown are we looking at? How does this affect the processor? I would appreciate some more information on how this works. Look here http://www.voip-info.org/wiki-Asterisk+cmd+ZapRAS for one option. or here for the other http://www.google.com/search?q=hdlc+zapata Hi Everyone, Thanks for all the input you add to the list. This seems to be a very good list. I am still new to Asterisk. If I run a PRI integrated T1 line into my office, do I need to split the line between the data and voice before plugging it into the asterisk box or is there some other way to do that? What are some good options for splitting the line? Thanks for any input. Tim BTW - Giving everyone a hug is an expression in Brazil. Everyone says it... it's like saying have a good one or good to see you. -- Steven Critchfield [EMAIL PROTECTED] ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.4 - Release Date: 3/18/2005 Paul Mahler www.signate.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
[Asterisk-Users] Asterisk locking up - 99.9% CPU
Are you at run level 3? X can cause this if you are at run level 5. Paul Paul Mahler [EMAIL PROTECTED] www.signate.com Hello We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work with our call agent. Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9% CPU. There is no debug output or other information that indicates there is a problem... Rather than continually restarting, can anyone make suggestions as to how we can track this down **OR** has anyone got the latest oh323/pwlb to work with CVS Head ? I see there is documentaiton on http://www.inaccessnetworks.com for the latest HEAD working with oh323 and pwlib... Any pointers would be appreciated -- Open WebMail Project (http://openwebmail.org) Paul Mahler www.signate.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] Any 24 (or 30) way FXS PCI cards?
I haven't used their 24 port gateway, only the four port gateway, but they have been excellent. http://www.mediatrix.com/products_devices.php?prodid=3 Paul Paul Mahler www.signate.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
[Asterisk-Users] Signate is now offering the dCAP test.
Now Available The Signate dCAP Certification Review and Exam for Asterisk--The Asterisk Open Source PBX Software Proficiency Certification. Signate, the leading provider of Asterisk-related training, will offer the dCAP Asterisk Professional Certification examination in San Francisco and other cities starting March 25, 2005, dCAP testing is offered the day after Signates three-day Introduction to Asterisk class. A half-day of training assists test takers by covering additional topics not covered in the Signate introductory course. To become a dCAP, candidates must pass a 150 question written test and a hands-on practical examination. The practical requires building and configuring a PBX. The $595 fee includes the $275 cost of the certification exam. dCAP certification will assure employers that they are talking to a qualified professional with proven proficiency in Asterisk PBX technologies, said Robert Messer, Founder of ABP Technology, a leading distributor of VoIP products through channel partners. We think our resellers will also benefit from the increased credibility that certification will bring them with prospective customers. Certified Asterisk Professionals receive many business benefits from certification including priority access to technical support, co-marketing opportunities, and the right to use the dCAP logo. Candidates attending the training with no prior hands-on Asterisk experience will find the dCAP examination difficult to pass. Signates Introduction to Asterisk and dCAP Certification Review prepares test takers for the certification exam. A Signate training class schedule is posted at http://www.signate.com/training.php. Certification is given under license from Digium. = Paul Mahler www.signate.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] Bluetooth phone as SIP handset?
I have a new HP IpaQ 6315. I run SJPhone on it with a bluetooth headset. Works great! Paul paul mahler www.signate.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Friday, March 04, 2005 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bluetooth phone as SIP handset? What head set are you using? We have the pro XTen and would like to be able to press a button on the BT device and pickup the call remotly. Just wear the BT on your ear as you walk about the office. You hear your softphone ring in your ear, press a button and Hello. -Matthew - Original Message - From: Linn Boyd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 04, 2005 1:11 PM Subject: Re: [Asterisk-Users] Bluetooth phone as SIP handset? Chris, I will take your $100.00 bounty :-D I am using a bluetooth headset with firefly and my laptop right now. Their softphone works well with asterisk. All you have to do is pair the headset to your computer, and set in the options to use the bluetooth. Mine works well. -Linn Chris Birkinshaw wrote: I know people are working on using a bluetooth phone as an extra line to send a receive calls through asterisk, but is anyone working on using a bluetooth phone as a handset - i.e. using it to dial calls and talk though asterisk? I would easily give upto $100 as a boounty for this functionality and I'm sure many others would too, as it would mean people wouldn't have to buy a hardware SIP phone or an ATA. Anyone know if this is possible? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.0 - Release Date: 3/2/2005 = Paul Mahler [EMAIL PROTECTED] ___ 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] High capacity voicemail - 5000 users isn't a lot
We supply an * server that can support as many users as you want, 5,000 is a small system. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alistair Cunningham Sent: Thursday, February 24, 2005 11:01 AM To: izo; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] High capacity voicemail Marcin, This depends enormously on the type of users. For 5000 users, I would normally recommend IBM's Unified Messaging product. For business users on that, I normally recommend at least one E1 per 2000 users as a starting point. For residential users, at least one E1 per 5000 users. IBM-UM is voicemail with web and email interfaces (which reduces traffic) and a lot of follow-me and operator services (which increases traffic if you can't do release link transfers, and reduces it if you can). Asterisk would be similar. If you can do release link transfers, then you can get more users per trunk; in some scenarios up to double the number. The above figures assume you can. The above figures are just a starting point; there is a good chance that your users are not typical. For example, in some countries in Asia, you can support many more business users - if callers hear a voicemail greeting, they usually hang up immediately, and call the person's mobile phone. You really need to do a pilot with a subset of users to gauge how much they use the system, and what features they use. For 5000 users, I would also consider some form of redundancy and automatic failover. 5000 angry users is not a pleasant sight! My company, Integrics Ltd, does consulting and installations of both Asterisk and IBM-UM. If you'd like help planning and installing such systems, drop me an email or give me a call. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ izo wrote: Hi, Does anybody has experience with high capacity PSTN voicemail and asterisk, running more then 5k mailboxes for PSTN users ? How many mailboxes can I serve with 4xE1 card if we assume that we have enough harddrive capacity. What would be server requirements. Would the CPU load be the same when storing voicemails in gsm format as compresing to gsm for ip calls ? Any hints would be greatly appreciated regards m. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 2/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 2/22/2005 ___ 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] redhat9 100% CPU
HI, 1. Make sure you are running asterisk with the command asterisk With no arguments. 2. Make sure you are booting to run level 3 so that X-windows isn't running. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TELUX Sent: Thursday, November 25, 2004 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] redhat9 100% CPU Redhat 9 is running 100% cpu usage. I had a couple boxes doing this. upgraded to Fedora and its ok. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QoS Router/Software Suggestions
The linksys BEFSR81 does QoS. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Tuesday, October 12, 2004 7:00 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] QoS Router/Software Suggestions I've got a Linksys BEFSR41 at home with RoadRunner service. I'm pretty sure it doesn't do QoS. I'm using WinXP Pro and not sure if it does QoS. I'm using SJ Phone and...(follow the pattern). I have to stop all network traffic on my machine if I want to have any hopes of making a clear call. But I shouldn't need to do that, right? Because somewhere the data packets should be getting queued and my voice packets should be having top priority right? How can I ensure this? Thanks, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Non-PRI T1 configuration
Are they just sending dnis? Do you have feature group D?
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, September 25, 2004 10:01 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Non-PRI T1 configuration
I'm trying to hook up a non-PRI fractional T1 using a T400P
port. The Telco says that it is provisioned as AMI with SF
(not ESF) and that they are signalling by sending down a
straight DS1 (I'm not sure what exactly that means). They
are also sending DNIS over these channels. I currently run it
through a channel bank for my IVR application and it works
fine but I am now trying to convert to *.
This leaves me with three questions. First, * does not have
an option for SF framing. If I use ESF, should that work or
is there another way?
Second, how do I configure the channel signalling in both
zaptel.conf and zapata.conf?
Third, how can I capture the DNIS in this situation or will
it automatically be available in the ${EXTEN} variable and
also passed to AGI scripts?
I would appreciate any help.
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RE: [Asterisk-Users] Asterisk and Red Hat 9
we run Asterisk on RedHat 9 with no problems. Works great! Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC665 Third StreetSuite 100San Francisco, CA94107-1901Asterisk Services and Training From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry DevitoSent: Sunday, September 19, 2004 2:30 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk and Red Hat 9 Hi everyone, Im a newbie to Asterisk. Will Asterisk run on RH9, easily or does it have to run on FreeBSD? Will the drivers for the Digium cards work on RH9? signate small logo.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler
Beginners. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sys. Concept Inc. Sent: Saturday, September 11, 2004 9:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] VoIP Telephony with Asterisk by Paul Mahler Does anybody have the book: VoIP Telephony with Asterisk by Paul Mahler. Is it for beginners or advanced users? -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linux distribution
Asterisk should run well with any Linux distribution. Mepis, www.mepis.org, is pre-configured for * and might make your installation faster andeasier. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC665 Third StreetSuite 100San Francisco, CA94107-1901Asterisk Services and Training From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Xavi CarolSent: Saturday, September 04, 2004 3:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Linux distribution Hello, Could anybody tell me if there is a Linux distribution (or Kernel version) that works better with Asterisk. I am newbie and I dont know if there is a preferred Linux/kernel version for Asterisk. Thanks. signate small logo.gif___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] which distro for asterisk?
The Mepis Debian distro is pre-configured for *, www.mepis.org They spent a lot of time making Mepis work with * out of the box. Everyone has their own very strong opinions on which distro is better. I'm not about to get into that. All I can say is Mepis is probably your fastest easiest way to get * running. You can get Linux installed and * running VERY quickly if you start with Mepis. Hope this helps, Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Tuesday, August 31, 2004 6:07 AM To: Asterisk Users List Subject: [Asterisk-Users] which distro for asterisk? Hi I want to play a bit with Asterisk. I currentlly install a new system for that and I would like to get your recommendations regarding the linux distro to use there. This is NOT intended to become a general distro flame war. My favorite distro is and no argument that you flame will convince me here (probably because I've heard it before). However I would like to minimize the OS maintinance task. I really wouldn't like to start worrying about upgrading sshd due to some stupid secuirty hole, and to worry what will it break on my system. I expect my distro to do that for me. I'd also like to have solid astrisk packages that won't break unnecessarily when the sshd package is updated next time. Hopefully also some sort of integration of zaptel in the distro's kernel package. I saw numerous complaints about unofficial RPM packages of asterisk. Besides them, the following free distros include asterisk packages: 1. Debian: http://packages.debian.org/asterisk . 2. Gentoo: Current package seems to be version 0.9.0 from 10-May-2004 3. The DAG repository for RH/Fedora: http://dag.wieers.com/packages/asterisk/ I have some experince with Debian, Mandrake and RedHat/Fedora. I'm unfamiliar with Gentoo and I have no good/bad experince with DAG packages with respect to quality and stability. Any recommendations, relevant experince and other learned opinions? thx -- Tzafrir Cohen +---+ http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend| mailto:[EMAIL PROTECTED] +---+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone recommendation for Receptionist
The expansion module is NOT supported with SIP. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Bogan Sent: Sunday, August 22, 2004 7:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone recommendation for Receptionist I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. What would you guys recommend? A Cisco 7960 with the 7914 expansion module [ http://www.cisco.com/en/US/products/hw/phones/ps379/ps1856/ind ex.html ] -- jeremy bogan[ [EMAIL PROTECTED] ] segment publishing - design.develop.host ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Call Redirection / Transfers
Under what circumstances? If the first T1 is down, for example? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Harragin Sent: Tuesday, August 03, 2004 7:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI Call Redirection / Transfers I have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one extention, multiple phones
You can easily ring different phones at the same time within the dial
command. For example,
SIP/4024${PRITRUNK1}/16505551212${PRITRUNK1}/1411212
A blind transfer will move the call to the next phone. Or you can park the
call.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean McKay
Sent: Saturday, July 31, 2004 5:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] one extention, multiple phones
Is it possible to get a few 7960's and asterisk to allow all
of the 7960 phones to use one extentsion and can only be used
by one person at a time, have it indicate on the other 7960's
when one of the others has the line engaged. Basicly so like
I can setup a rule when an incoming call comes from IAX to
divert to this extension, it will ring the extension (thus
all phones), and allow me to place a call on hold on one
phone and pick it up on another and the original phone would
acknowledge that the call has been picked up and disengage.
Can I do this without call parking?
Basicly the same model as having a bunch of phones on a pstn
line with each phone having a hold button. The goal here is
to allow me to pick up a call on any of the 7960's anywhere
in my house and be able to move from room to room as needed
by placing the call on hold and picking it up on one of the
other phones in the house.
/\ . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
. . . . . .
\ / - ASCII Ribbon Campaign . Sean McKay - [EMAIL PROTECTED] X -
NO HTML/RTF in e-mail . Team Lead, bahamut web team / \ - NO
Word docs in e-mail . ircd-qa team member
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RE: [Asterisk-Users] Re: Really long first ring, then normal
I have a recent version installed. I am having problems with hangup detection on my zap channels. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: Friday, July 16, 2004 5:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Really long first ring, then normal Hiya, I've been seeing this lately with our Asterisk PBX as well. With Asterisk CVS HEAD 05-21, the ringing was normal. After our last upgrade, CVS HEAD 07-03, the first ring is almost always one continuous ring that lasts about 2 to 3 seconds. We're noticing other problems too, such as hangup detection. This worked flawlessly for us with CVS HEAD 05-21 (we have a TDM400P card with 1 FXS and 1 FXO). But, now Asterisk isn't detecting hangups at all. I'm not sure at the moment if something on the actual POTS line has changed, or if it's a problem with the CVS version we're running. Anyone else noticing strange behaviour such as the above? jup, me2: If i'm calling my Cisco 7960 from my cellphone, the Cisco rings 3-5 times before i hear the ring on the cellphones. I tried progress=yes and no in sip.conf, no difference. The calls come in on a BRI with chan_capi, OR on a Cisco 3620 with VIC-2BRI, it's the same problem both ways, so its neither capi nor a SIP problem... I'm using CVS from today. I've just opened a bug with ID2062, maybe someone knows when exacty this problem started... Andreas. _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Sunday, July 11, 2004 1:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I explained him a sample need. I don't think asterisk does whatever i want in sip. It is an good PBX. Please help me to understand. Anywhere am I wrong ? Or as you say is that SIP feature is written? -Kannaiyan. - Original Message - From: usedcanon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 10:02 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I was going to keep out of this (was interesting to read, as I have dealt with simmillar situation) however I would like to add just this one commnet. Try to better understand asterisk than to throw about your money. What you want to do is perfectly possible with asterisk there is no need to add a new confusing feature. As for your bounty, donate it to the wiki ! :-) Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan Sent: 11 July 2004 09:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I accept your views. I have a specific requirements, can you help to attain the same. In our business we have 25 employees handling customer service. I want to add or remove employees in the customer service so does the devices connected to it. I don't want to make any changes in the asterisk, and all I need is to plug in the VoIP Phone and start handling the customer service. I would like to do for as many employees as I want without any problems. Can you think of a better solution? -Kannaiyan. - Original Message - From: Sunrise Ltd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:15 AM Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for that is the dial command. However, there is no implicit assignment between an extension and an account and that is good so. This should not be changed because it would harm Asterisk's flexibility and manageability. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing As I said, you are confusing extensions with accounts. The first 12345 is an extension, the three (12345)s are accounts. Those are different layers, don't mix them up. You should always be able to distinguish between devices, even if they are assigned the same phone number. In fact, in a call centre you'd be using a call queue. It would be rather nonsensical for a call queue management to have to distinguish between multiple identical agents. Therefore, setting up multiple devices with the same account credentials is not a good idea, especially not in a call centre. Each device and each agent should have their own unique account credentials and assigning extensions to them should always be done through the dialplan and only the dialplan. Asterisk has been designed this way. It is a good design. It should NOT be changed nor undermined. You may want to do something like this ... [GLOBALS] A-GROUP = SIP/2001 SIP2002 SIP/2003 B-BROUP = SIP/jdoe SIP/dflint SIP/bsmith ... [Support
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Sunday, July 11, 2004 9:57 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous On 11/07/2004 at 08:42 Paul Mahler wrote: You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. Sorry, but this is irrelavant... SIP allows multiple endpoints to register with the same account details and will all ring when called. The fact that the rtp stream goes to the first endpoint to pick up (and respond) is what's important ie, if multiple devices are registered with the same account they will *all* be 'spoken' to... Asterisk currently does not support this behaviour. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. No, the idea is to get asterisk to act like a real sip proxy. The dialplan solution is a poor hack. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. again, irrelavant - the whole beauty of the way SIP works is that I can add to the list of phones that get called by simply registering more phones with the same details. I don't need my users to mess with or make a support call to add to the dial plan. They can add and remove themselves. I'd also suggest adding something like registrationlimit=1 for those that do not want to support multiple client registrations, I'd also like to see the implementation of the q parameter... I'm all for this modification to SIP, although I'd probably want to see DTMF callerid implemented first :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
It's not what SIP does with SER, it's what SER does with SIP. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Sunday, July 11, 2004 9:58 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous As Daniel Says, Bounty stands. I cannot explain to you anymore. I'm sorry. Please read more what SIP can do with SER. -Kannaiyan. - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 4:42 PM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous You are confused about what a SIP session is and what a SIP session does. SIP, session initiation protocol, controls an RTP, real time protocol, session between two IP endpionts. The end points have to have unique IP addresses for the session to run. The unique SIP registration is how * finds a UNIQUE endpoint. You don't want SIP to solve your problem, you want * to solve your problem. You are asking for this SIP feature because you are confused as to how SIP and * work, and how they work together. You can easily fix your business problem with *, but not with mechanism you are asking for. You should spend your money on getting a copy of each of the two books that are now available and learn *. Then it will be clear to you that you don't really want what you are asking for. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kannaiyan Natesan Sent: Sunday, July 11, 2004 1:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I explained him a sample need. I don't think asterisk does whatever i want in sip. It is an good PBX. Please help me to understand. Anywhere am I wrong ? Or as you say is that SIP feature is written? -Kannaiyan. - Original Message - From: usedcanon [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 10:02 AM Subject: RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I was going to keep out of this (was interesting to read, as I have dealt with simmillar situation) however I would like to add just this one commnet. Try to better understand asterisk than to throw about your money. What you want to do is perfectly possible with asterisk there is no need to add a new confusing feature. As for your bounty, donate it to the wiki ! :-) Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan Sent: 11 July 2004 09:51 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous I accept your views. I have a specific requirements, can you help to attain the same. In our business we have 25 employees handling customer service. I want to add or remove employees in the customer service so does the devices connected to it. I don't want to make any changes in the asterisk, and all I need is to plug in the VoIP Phone and start handling the customer service. I would like to do for as many employees as I want without any problems. Can you think of a better solution? -Kannaiyan. - Original Message - From: Sunrise Ltd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:15 AM Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous When I call a SIP user, the phone should ring in more than one extentions. Also more than one phone should be able to register with asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for that is the dial command. However, there is no implicit assignment between an extension and an account and that is good so. This should not be changed because it would harm Asterisk's flexibility and manageability. This type of situations might be needed in call centres. Called 12345 |---(12345) Ringing |---(12345) Ringing |---(12345) Ringing As I said, you
RE: [Asterisk-Users] QoS in asterisk
This is a very complex question. First, you have to ask about VoIP and QoS. This is because * uses VoIP protocols like UDP and RTP. In general, the QoS of VoIP is not as high as with the PSTN. Even so, call quality can be generally very good. Second, * does support features that support QoS, for example the IAX jitterbuffer setting. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] QoS in asterisk
Well, the question may not have been about QoS, but my answer certainly was. QoS is defined as The performance specification of a communications channel or system. (188) Note: QOS may be quantitatively indicated by channel or system performance parameters, such as signal-to-noise ratio (S/N), bit error ratio (BER), message throughput rate, and call blocking probability. Quality of Service (QoS) is a general term for an abstraction covering aspects of the non-functional behavior of a system, for example delay. I think what we have here is what we are going to see a lot of--cultures in collision. The PSTN folks had QoS issues long before it became an IP issue. I think what you are alluding to is routing specific IP QoS. IP supports QoS in the IP header, those pesky tos bits you were talking about. Asynchronous transfer mode (ATM) natively provides QoS. The IEEE 802.1p standard covers QoS in all IEEE 802-type networks. Even in networking, QoS is first and formost an abstraction before it becomes a specification. QoS is the ability of a network element (e.g. an application, a host or a router) to provide some level of assurance for consistent network data delivery. QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonble question to me. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen J. Wilcox Sent: Sunday, July 11, 2004 3:33 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] QoS in asterisk Both the question and the answer are not talking about QoS. From the Q, qos does not provide a measure of quality, it provides a system to allow you to request your data be handled according to priorities. From the A, qos is confused with the pstn.. qos is a feature of IP, that has nothing to do with the pstn. jitterbuffer isnt qos either, altho its important you get it right to provide good quality calls. qos is the tos options you can specify in the conf files but you need to combine that with routers from server to client that will honor the tos you set. deciding to switch from voip to pstn wouldnt be covered by qos, you would need to find other ways.. playing with options like 'qualify' to timeout poorly connected devices quickly is more like what you are trying to achieve. Steve On Sun, 11 Jul 2004, Paul Mahler wrote: This is a very complex question. First, you have to ask about VoIP and QoS. This is because * uses VoIP protocols like UDP and RTP. In general, the QoS of VoIP is not as high as with the PSTN. Even so, call quality can be generally very good. Second, * does support features that support QoS, for example the IAX jitterbuffer setting. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Does asterisk provide quality of service(QoS)? If it does, how do I use it? The reason why I ask is that I need to switch to use POTS should the internet connection becomes poor? Thanks, Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Well, this is certainly getting exciting. Andy, I took your advice and re-read the RFP. Andy--I don't think you are a good candidate for a beginner's book on *, but if you send my your address, I'll send you a copy on me. :-) So, gentlemen, help me out here. The spec says: The Address of record is the SIP address that the registry knows the registrand. . . A Sip message is either a request from a client to a server or a response from a server to a client. A client uses the REGISTER method to register the address listed in the TO header field with a SIP Server. And as Nick so cogently pointed out Once a client has established bindings at a registrar, it MAY send subsequent registrations containing new bindings or modifications to existing bindings as necessary. The 2xx response to the REGISTER request will contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record at this registrar. I don't see how two different clients can register with a server as the same address of record. Doesn't the second registration from a new client change the address of record for the registered client? If the second client is trying the same registration as the first client, and it's the responsibility of the client to provide the complete list of bindings, how does the second client know the list of bindings for the first client that bound the registration? So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? Andy, I'm in your hands. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicholas Bachmann Sent: Sunday, July 11, 2004 12:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous Mike Machado wrote: On Sun, 2004-07-11 at 12:31, Paul Mahler wrote: The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. Did you even read the RFC? Section 10.2.1 clearly talks about adding multiple bindings to the same address-of record. Just to quote and save everybody the searching: Once a client has established bindings at a registrar, it MAY send subsequent registrations containing new bindings or modifications to existing bindings as necessary. The 2xx response to the REGISTER request will contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record at this registrar. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Book
Not from me. I think the more books the better. I'm looking forward to getting my copy. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Bailey Sent: Friday, July 09, 2004 2:52 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Book Hello, If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/ I ordered a copy, but they said it's six weeks or so 'till delivery. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training Do I detect some friendly rivalry? ;-) | VoIP Telephony with Asterisk will be available July 22, directly from | Signate and through selected resellers for $49.95 plus shipping. Call | 415-442-4011 to order the book. Seriously, though, the more documentation the better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] German Asterisk Site
Das is aber schöen! Paul von Wachter Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jo Sent: Saturday, July 10, 2004 8:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] German Asterisk Site Beierlein Moritz wrote: Hello Asterisk Users, is there a good german site for asterisk? Moritz Hi Moritz, there is * dicussion group at the German IP-Phone forum: http://www.ip-phone-forum.de/ http://www.ip-phone-forum.de/forum/viewforum.php?f=24 jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Three (quick?) questions...
Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry
I'm not sure I understand what you are trying to do. You have an administrative assistant and several other staff. You want the administrator to be able to take calls directed to the staff extensions? If I have the requirement right, you could accomplish this by ringing the staff extension and the admin extension at the same time. The Dial command allows you to ring multiple extensions simultaneously. If you want to be able to more easily recognize what extension the traffic if for, you can add additional extensions to the 7960. For example, if you have two staff the admin monitors, add two additional extensions to the 7960. The admin can tell who is being called by the extension that rings. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Saturday, July 10, 2004 3:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+s imultaneous+registry Updated, Allow a SIP device to register more than once so a single extension may exist in multiple locations. Upped total to $75. Daniel... Daniel Jimenez wrote: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultane ous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Book
I ordered a copy, but they said it's six weeks or so 'till delivery. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of pat munis Sent: Thursday, July 08, 2004 11:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Book does anyone know anything about this Book? - Original Message - From: [EMAIL PROTECTED] Date: Mon, 5 Jul 2004 00:15:44 +0600 (MDT) To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Book If anyone is interested in getting a book on asterisk I would recommend checking out http://www.saww.net/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Sign-up for Ads Free at Mail.com http://promo.mail.com/adsfreejump.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Firefly release - 1.9.3
Ok, you asked for it, so here it is. ;-) Fabulous! Works great! Love Firefly! Magnificant job! Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: Monday, June 28, 2004 9:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Firefly release - 1.9.3 There's a new firefly release out for those who are using firefly with your lovely asterisk / SIP server. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe the main changes are improved GUI fixes (mouse wheel works now :) ), few url parsing fixes, mic volume control and improved compatibility with SIP servers (namely SER). Send me all bugs, problems and suggestions (even praise if you're feeling generous) -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stable branch usable? Development branch better?
Is the stable branch usable? Is there ever going to be a 1.0 release? Should I be using the "stable" branch or the development branch? The development branch seems to have more fixes than the stable branch. It looks like fixes going into the release branch aren't going into the stable branch. TKS Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC665 Third StreetSuite 100San Francisco, CA94107-1901Asterisk Services and Training signate small logo.gif
RE: [Asterisk-Users] IAX2 Trunking help!
The register statement informs a remote * server of the location of the local * server. If the local server is always at the same IP address, there is no need to register with the remote server. IAX will work just fine with no register statements at either side. Here is a typical iax.conf configuration to allow incoming connections from a remote * server. This is from a working installation. ; Inter-Asterisk eXchange driver definition (asterisk2) [general] port=4569 accountcode=lss0101 bandwidth=low allow=g723 allow=g729 disallow=lpc10 allow=gsm tos=lowdelay [asterisk] type=friend auth=md5 secret=1945 context=local host=dynamic defaultip=10.1.1.180 qualify=yes This statement at the remote * server would dial the local server with the configuration shown above. Incoming calls would be directed to the local context in extensions.conf. exten = _1XXX,1,Dial(IAX2/asterisk:[EMAIL PROTECTED]/[EMAIL PROTECTED]) Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Cook Sent: Tuesday, June 22, 2004 5:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 Trunking help! I was just trying to solve this one myself. I found this method worked for me. I'm still calling this Method 1 in my document because I don't fully understand the switch and the register versions and pros/cons to implementation of each. But this one does work. Method 1 Receiving Server Iax.conf [REC_SERVER] type=user host=my.calling.server.ca secret=mysecret context=local trunk=yes Calling Server Extensions.conf [mycontext] exten = _5XXX,1,Dial(IAX2/REC_SERVER:[EMAIL PROTECTED]/$ [EMAIL PROTECTED]) exten = _5XXX,2,Hangup exten = _5XXX,102,Hangup Any call in the mycontext context on Calling Server to extensions 5000-5999 (mapped by extension _5XXX) will get sent to receiving server (my.receiving.server.ca) into the local context on the receiving server. Performing the same configuration in the opposite direction will allow cross-calls between Asterisk systems. -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CISCO 7960 Goes missing
Telnet to the phone and look at the sip debug trail. Probably a wrong ip address somewhere. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 22, 2004 6:45 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CISCO 7960 Goes missing I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:43:12 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:43:36 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! I've checked the cable and even swapped out the phone but 4001 is always disappearing off of the network. Anyone got any hints? Obviously I've added qualify=yes to my sip.conf in an attempt to troubleshoot this but I'm now getting nowhere fast! Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CISCO 7960 Goes missing
You could also install ngrep and watch the traffic go by on port 5060. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 22, 2004 6:45 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CISCO 7960 Goes missing I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:43:12 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:43:36 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! I've checked the cable and even swapped out the phone but 4001 is always disappearing off of the network. Anyone got any hints? Obviously I've added qualify=yes to my sip.conf in an attempt to troubleshoot this but I'm now getting nowhere fast! Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP error 407 - can't make outgoing calls
I am using a IPDialog siptone II. I can take incoming calls, but when I try and make an outgoing call I get a SIP 407 error. Can some kind soul explain to me what I am doing wrong? Here's what I found in the wiki: If a proxy does not accept the credentials sent with a request, it SHOULD return a 407 (Proxy Authentication Required). The response MUST include a Proxy-Authenticate header field containing a (possibly new) challenge applicable to the proxy for the requested resource. Here's what I have in sip.conf [514] type=friend ; This device takes and makes calls username=514 secret=password context=inside callerid=Paul Mahler 4154424024 qualify=1000 host=dynamic ; This host is not on the same IP addr every time canreinvite=no [EMAIL PROTECTED] ; Activate the message waiting light for waiting messages ;defaultip=192.168.0.102 Here's the sip debug showing the error: to 209.234.100.68:5060 Retransmitting #5 (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.234.100.68:5060 From: 514 sip:[EMAIL PROTECTED];tag=3397-f0f0c367 To: 503 sip:[EMAIL PROTECTED];tag=as0528d61b Call-ID: [EMAIL PROTECTED] CSeq: 14057 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=230958ab Content-Length: 0 The password at the phone is the same as the password in sip.conf. Thanks! Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compiling problem on Debian
If you use the mepis debian release, it's a piece of cake to install *. It takes about 15 minutes to install Mepis and *. www.mepis.org Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robin Calmegård Siurua Sent: Thursday, June 17, 2004 1:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Compiling problem on Debian Hi, I can't compile Asterisk on a Debian machine. gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o -ldl -lpthread -lncurses -lm -lresolv editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a qeditline/libedit.a(editline.o_a)(.text+0x7e6a): In function `term_move_to_line': /home/robin/asterisk-0.9.0/editline/term.c:554: undefined reference to `tgoto' editline/libedit.a(editline.o_a)(.text+0x7e7e):/home/robin/ast erisk-0.9.0/editline/term.c:554: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x7ef7):/home/robin/ast erisk-0.9.0/editline/term.c:567: undefined reference to `tgoto' editline/libedit.a(editline.o_a)(.text+0x7f0b):/home/robin/ast erisk-0.9.0/editline/term.c:567: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x7f40):/home/robin/ast erisk-0.9.0/editline/term.c:572: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x7fc0): In function `term_move_to_char': /home/robin/asterisk-0.9.0/editline/term.c:607: undefined reference to `tgoto' editline/libedit.a(editline.o_a)(.text+0x800e):/home/robin/ast erisk-0.9.0/editline/term.c:611: undefined reference to `tgoto' editline/libedit.a(editline.o_a)(.text+0x80bc):/home/robin/ast erisk-0.9.0/editline/term.c:643: undefined reference to `tgoto' editline/libedit.a(editline.o_a)(.text+0x80d0):/home/robin/ast erisk-0.9.0/editline/term.c:643: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x824c): In function `term_deletechars': /home/robin/asterisk-0.9.0/editline/term.c:734: undefined reference to `tgoto' editline/libedit.a(editline.o_a)(.text+0x8289):/home/robin/ast erisk-0.9.0/editline/term.c:739: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x82be):/home/robin/ast erisk-0.9.0/editline/term.c:743: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x82f0):/home/robin/ast erisk-0.9.0/editline/term.c:746: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x836a): In function `term_insertwrite': /home/robin/asterisk-0.9.0/editline/term.c:775: undefined reference to `tgoto' editline/libedit.a(editline.o_a)(.text+0x837e):/home/robin/ast erisk-0.9.0/editline/term.c:775: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x83d9):/home/robin/ast erisk-0.9.0/editline/term.c:782: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x8417):/home/robin/ast erisk-0.9.0/editline/term.c:790: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x8435):/home/robin/ast erisk-0.9.0/editline/term.c:792: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x8463):/home/robin/ast erisk-0.9.0/editline/term.c:797: undefined reference to `tputs' editline/libedit.a(editline.o_a)(.text+0x849e):/home/robin/ast erisk-0.9.0/editline/term.c:805: more undefined references to `tputs' follow editline/libedit.a(editline.o_a)(.text+0x86e2): In function `term_set': /home/robin/asterisk-0.9.0/editline/term.c:911: undefined reference to `tgetent' editline/libedit.a(editline.o_a)(.text+0x87d6):/home/robin/ast erisk-0.9.0/editline/term.c:929: undefined reference to `tgetflag' editline/libedit.a(editline.o_a)(.text+0x87ea):/home/robin/ast erisk-0.9.0/editline/term.c:930: undefined reference to `tgetflag' editline/libedit.a(editline.o_a)(.text+0x87ff):/home/robin/ast erisk-0.9.0/editline/term.c:932: undefined reference to `tgetflag' editline/libedit.a(editline.o_a)(.text+0x8811):/home/robin/ast erisk-0.9.0/editline/term.c:933: undefined reference to `tgetflag' editline/libedit.a(editline.o_a)(.text+0x8823):/home/robin/ast erisk-0.9.0/editline/term.c:935: undefined reference to `tgetflag' editline/libedit.a(editline.o_a)(.text+0x8835):/home/robin/ast erisk-0.9.0/editline/term.c:936: more undefined references to `tgetflag' follow editline/libedit.a(editline.o_a)(.text+0x8847): In function `term_set': /home/robin/asterisk-0.9.0/editline/term.c:938: undefined reference to `tgetnum' editline/libedit.a(editline.o_a)(.text+0x8859):/home/robin/ast
RE: [Asterisk-Users] New Firefly version
I'm having this problem too. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 7:53 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New Firefly version Why all the time the firefly show me the message: Sip registration failed for the network Home (407). The server, username and password are correct. I'm using the default RTP port 5000 in the SIP tab. Using the SJPhone I can register; using the firefly I can call any registered number, but I can't register. On asterisk console: Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 User-Agent: Firefly Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-436999556-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=5a1c4f36 To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 SAMPLANET1*CLI Sip read: REGISTER sip:192.168.199.3:5060;transport=udp SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;transport=udp From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER Contact: sip:[EMAIL PROTECTED]:5060 Expires: 3600 Max-Forwards: 70 Proxy-Authorization: Digest username=2003,realm=asterisk,nonce=38165263,uri=sip:192.1 68.199.3:5060; transport=udp,response=ec0afc0a2b13a725aa40b5c311c396d8,alg orithm=MD5 User-Agent: Firefly Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.199.121 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.199.121:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.199.121:5060;branch=z9hG4bK-c87542-373911025-1--c87542-;rport From: sip:[EMAIL PROTECTED]:5060;tag=6c3de14a To: sip:[EMAIL PROTECTED]:5060;transport=udp;tag=as7843e908 Call-ID: c90fa011e82acf3e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=38165263 Content-Length: 0 to 192.168.199.121:5060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] seeking an example for Message Waiting Indicator stutter dialtone
does anyone have an example they would please share for turning on stutter dialtone for a zaptel channel when there is a message waiting? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Receptionist manager program.
Please count me in for testing! Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Friday, May 28, 2004 8:33 AM To: Asterisk Subject: [Asterisk-Users] Asterisk Receptionist manager program. We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. We are considering writing a SIP client build into the program at a later time. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No stutter MWI on zaptel channel with message waiting
This one is making me crazy. I have a T1 connection from * to an adit channel bank. When there is voicemail, there is not stutter for message waiting indicator. Here is zapata.conf. Thanks! [channels] language=en signalling=fxo_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes ; spans one and two connect to the Adit 600 channel bank signalling=fxo_ks group=1 context=inside channel = 1-48 callgroup=1 pickupgroup=1 callerid=Paul Mahler 100 context=inside [EMAIL PROTECTED] channel = 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help! No stutter dialtone on message waiting - zaptel phones
I have the following entry in zapata.conf, but I don't get stutter dialtone when there is a message waiting. Suggestions? Please? callgroup=1 pickupgroup=1 callerid=Paul mahler 100 context=inside mailbox=100 channel = 1 Thanks, Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 100 analog phones?? HOWTO?
I have had good experiences with Adit. Their customer service and documentation are excellent. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Gustafson Sent: Monday, May 24, 2004 4:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 100 analog phones?? HOWTO? Does anyone know the best approach to take for handling 100 analog phones? It seems to me that a chassis like Carrier Access or Adtran would work. The chassis would do much of the hard work of converting the analog sound to data. Any recommendations on hardware for the chassis? ...Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP!!! How do I move voicemail files to a new machine?
I copied voicemail files to a replacement system. When vm tries to play the file * throws an error messages: Unexpected header size 16 unable to open fd on / How can I copy the VM to the new machine? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HOW do I restore voicemail from backups?
I am trying to recreate an * server from backups. I copied /var/spool/asterisk/voicemail/context/109/INBOX/* from backups. The voicemail files got restored msg.gsm msg.txt msg.wav but when the user goes into voicemail, * says there is no voicemail. Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ADIT 600 Manual
It's avaialble at: http://www.carrieraccess.com/support/products/index.cfm/fuseaction/default_p rod/cat_id/21.htm Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Brandon Sent: Tuesday, May 18, 2004 4:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ADIT 600 Manual I am trying to find a manual for the Carrier Access Adit 600. Does anyone know where I might be able to find one? Thanks -Jon -- Jon J. Brandon[EMAIL PROTECTED] http://www.monsoonretail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Free Softphone Recomendations
Title: Message Does the Cisco softphone work with SIP? The factsheet only talks about SKINNY. Paul Mahler [EMAIL PROTECTED] Signate, LLC665 Third StreetSuite 100San Francisco, CA94107-1901Asterisk Services and Training From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listsSent: Tuesday, May 18, 2004 6:44 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Free Softphone Recomendations humm now that I think about I don't think it's free sorry my mistake -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of listsSent: Tuesday, May 18, 2004 9:27 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Free Softphone Recomendations what about cisco's Ip comunicator? it's free so is the old cisco soft phone. If you don't have access to it let me know Doug BlockChief Information Officer of Efast Funding713-983-4055 (Direct)888-338-3863 x 4055 (Toll Free)713-983-4555 (Direct Fax) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron MartinSent: Tuesday, May 18, 2004 8:55 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Free Softphone Recomendations Does anyone have any recomendations for a free Windows softphone, SIP or IAX that supports the following features: * Message Waiting Indicator * Consultative Transfers * Speed Dials signate small logo.gif
RE: [Asterisk-Users] What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Excellent answer. Thank you very much.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andreas Frackowiak
Sent: Saturday, May 15, 2004 1:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] What's in ${EXTEN} ? Why does
voicemail prompt for an extension?
Why does voicemail prompt me for an extension instead of
just asking
my password?
Because there is no Voicemailbox 99 in that context in your
configuration.
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
In your example, $EXTEN will always be 99, because that is
the extension.
If you would like to have the 99 as a prefix for the
following voicemailbox number you could do something like:
exten = _99.,1,VoicemailMain(${EXTEN:[EMAIL PROTECTED])
exten = _99.,2,Hangup
And then 99123 would go directly to Mailbox 123 (if it exists).
regards
Andreas
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[Asterisk-Users] How to Echo extension number to caller?
I need to dial an extension that tells me what extension I'm dialing from.
I'm running a bunch of analog phones off a channel bank to * over a T1. I
have the following in extensions.conf.
exten = 98,1,SayDigits(${EXTEN})
This says the digits the caller enters on the keypad, not the extension they
are calling from.
Thanks Guys
Paul
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.signate.com/
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
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[Asterisk-Users] What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten = 99,1,VoicemailMain([EMAIL PROTECTED]) exten = 99,2,Hangup Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Business Looking Analog Phone
Cylogistics sells a sayson phone that's very nice. http://cylogistics.comtelligence.net http://www.sayson.com/product/analog_phone.htm Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, May 14, 2004 8:09 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Business Looking Analog Phone I am looking for some analog phones are low cost yet not cheap looking. they should fit into a business setting. Can anyone help? Aastra PT390s. Kick ass phones. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Terrible TICKING sound - Fixed
Using the sunc from the T1 line made my problems go away. Thanks Andrew!!! Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, May 11, 2004 3:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Terrible TICKING sound The Adit channel bank we are using, and XO communications who provisioned the T1 are both showing a LOT of framing errors on our system. Tell Asterisk to clock from XO's T1. How is this related to your TDM400P though? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Terrible TICKING sound
The Adit channel bank we are using, and XO communications who provisioned the T1 are both showing a LOT of framing errors on our system. Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tmpm Sent: Tuesday, May 11, 2004 12:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Terrible TICKING sound Ive found this in audio apps on other boxes when the power supply is really loaded down hard. Just one more maybe for you to check. Have you blown the dust out of the P/S lately? Dirt and temp variations seem to affect it as well...found this with audio equipment at a broadcast station. (Streaming server on a Linux box) Not saying its your situation, but wont hurt to check. If it does help, a beefier power supply might help here. It cured my case. Marc At 10:45 5/11/2004, you wrote: I've fought with this problem on and off. Number 1 thing to check is /proc/interrupts to ensure that your card isn't sharing an interrupt with something else. Number 2 is a bit of an unknown variable - my guess is either electrical noise, or perhaps vibrations affecting your card inside your box. I find that carefully remounting my tdm400p/x100p so that nothing at all is touching it (no wires, no plastic, nothing - except at the mount point) will make the problem go away the majority of the time. If it doesn't go away, try re-mounting again. It's a little scary, especially when you're working with a small-form-factor machine. Ryan On 10-May-04, at 8:43 PM, Paul Mahler wrote: i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to record all agent calls
you need to combine both sides of the conversation. This should be covered in the archives. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLCPO Box 60430Palo Alto, CA94306VoIP Systems, Training Consulting From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff CrewsSent: Tuesday, May 11, 2004 2:57 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] how to record all agent calls I want to record incoming calls that are queued when the call is connected to an agent.I added the following lines to agents.conf before the list of agents:; Enable recording calls addressed to agents. It's turned off by default.recordagentcalls=yes;;The format to be used to record the calls;wav, gsm, wav49.; By default its "wav".recordformat=gsm;; Insert into CDR userfield a name of the the created recording; By default it's turned off.createlink=no;; The text to be added to the name of the recording. Allows forming a url link.;urlprefix=http://host.domain/calls/;; The optional directory to save the conversations in. The default is; /var/spool/asterisk/monitor;savecallsin=/var/callsand added to the queues.conf file:; monitor-format = gsm|wav|wav49monitor-format = gsm...and then issued the reload command in the Asterisk CLI console.I even created the /var/log/asterisk/monitor directory because it did not exist.Is there something else that needs to happen to record calls between agents and callers so you can hear both sides of the conversation?Thanks in advance. ---Jeff CrewsEastern Oregon Net, Inc.La Grande OregonEmail [EMAIL PROTECTED]Voice 541-963-2625 or 800-785-7873, extension 11 personal efax 503-907-6704, standard company fax 541-962-7818 web http://home.eoni.com signate small logo.gif
RE: [Asterisk-Users] Terrible TICKING sound - Fixed
Well, for me it was a problem with the T1 line. XO fixed the line and the ticking sound is gone! Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, May 11, 2004 3:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Terrible TICKING sound The Adit channel bank we are using, and XO communications who provisioned the T1 are both showing a LOT of framing errors on our system. Tell Asterisk to clock from XO's T1. How is this related to your TDM400P though? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Terrible TICKING sound
i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.signate.com/ Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WI FI IP phones??
I guess vocera doesn't have any RF engineers to tell them they can't do it. Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Musone Sent: Friday, May 07, 2004 9:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] WI FI IP phones?? Why not vocera? http://www.vocera.com they seem to have the exact product you are looking for and seem to primarily server hospitals.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Moran Sent: Friday, May 07, 2004 1:06 PM To: Asterisk Subject: Re: [Asterisk-Users] WI FI IP phones?? Hmm I'll look into it. Thanks. On Fri, 2004-05-07 at 12:54, John Fraizer wrote: James Moran wrote: No I'm not but it's a hospital that nurses are on call and need to have a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote: James Moran wrote: We need to have about 30 phones on one floor And you really think that WiFi phones are suited for this application? Not an RF engineer, are ya? John Um, I'm not so sure that you're going to be able to run WiFi at a hospital. The life safety/support equipment is most likely not certified to be resistant to 2.4Ghz interference. It's been a while since I looked up ISM allocations but, I can tell you that I've seen many good ideas shot down because of the potential to interfere with the medical equipment. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Moran [EMAIL PROTECTED] Potential Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail or voicemail2?
I'm using the stable branch. Is voicemail or voicemail2 deprecated? TKS Paul [EMAIL PROTECTED]
[Asterisk-Users] Why don't I get a ringing sound?
I am using the following macro to dial a ZAP channel. When I dial in, *
answers and I go to voicemail. I never hear any ringing, though. It doesn't
work with the Ringing command before or after the Dial command.
[macro-zapdial]
;
; call a ZAP extension for ${ARG2} seconds, and then voice mail
; ${ARG1} - Extension
; ${ARG2} - Time to ring
exten = s,1,Dial(ZAP/${ARG1},${ARG2})
exten = s,2,ringing
exten = s,3,Voicemail(u$[${ARG1} + 99]) ; match the channel to the mailbox
exten = s,4,Goto(${ARG1},1) ; If they press #, return to start
exten = s,104,Voicemail(b$[${ARG1} + 99])
exten = s,5,Goto(${ARG1},1) ; If they press #, return to start
Here is what the log shows:
-- Zap/1-1 is ringing
-- Nobody picked up in 2 ms
-- Hungup 'Zap/1-1'
-- Executing Ringing(Zap/49-1, ) in new stack
-- Executing VoiceMail(Zap/49-1, u100) in new stack
-- Playing 'vm-theperson' (language 'en')
May 2 18:36:45 WARNING[163851]: chan_zap.c:1193 zt_set_hook: zt hook
failed: Device or resource busy
-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (macro-zapdial, s, 3) exited non-zero on 'Zap/49-1' in
macro 'zapdial'
== Spawn extension (main, 100, 1) exited non-zero on 'Zap/49-1'
-- Hungup 'Zap/49-1'
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Why don't I get a ringing sound? - DUH!
I got it! Nothing like posting to the mailing list when you're going to look
stupid to help you find the answer yourself!
The answer is to use waitforring(1)!
Thanks!
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Sunday, May 02, 2004 2:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Why don't I get a ringing sound?
I am using the following macro to dial a ZAP channel. When I dial in, *
answers and I go to voicemail. I never hear any ringing, though. It doesn't
work with the Ringing command before or after the Dial command.
[macro-zapdial]
;
; call a ZAP extension for ${ARG2} seconds, and then voice mail
; ${ARG1} - Extension
; ${ARG2} - Time to ring
exten = s,1,Dial(ZAP/${ARG1},${ARG2})
exten = s,2,ringing
exten = s,3,Voicemail(u$[${ARG1} + 99]) ; match the channel to the mailbox
exten = s,4,Goto(${ARG1},1) ; If they press #, return to start exten =
s,104,Voicemail(b$[${ARG1} + 99]) exten = s,5,Goto(${ARG1},1) ; If they
press #, return to start
Here is what the log shows:
-- Zap/1-1 is ringing
-- Nobody picked up in 2 ms
-- Hungup 'Zap/1-1'
-- Executing Ringing(Zap/49-1, ) in new stack
-- Executing VoiceMail(Zap/49-1, u100) in new stack
-- Playing 'vm-theperson' (language 'en') May 2 18:36:45
WARNING[163851]: chan_zap.c:1193 zt_set_hook: zt hook
failed: Device or resource busy
-- Starting simple switch on 'Zap/1-1'
-- Hungup 'Zap/1-1'
-- Playing 'digits/1' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'digits/0' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (macro-zapdial, s, 3) exited non-zero on 'Zap/49-1' in
macro 'zapdial'
== Spawn extension (main, 100, 1) exited non-zero on 'Zap/49-1'
-- Hungup 'Zap/49-1'
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
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[Asterisk-Users] Is SIP BROKEN?
in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here [globals] [inside] exten = 77,1,voicemailmain [other] exten = 88,1,Playback(demo-congrats) Next, I have an x-lite phone set up as Display name: 40 Username: 40 Authorization user: 40 Domain/Realm: 69.240.152.95 SIP Proxy: 69.240.152.95 I get a message from SIP debug that says 40 from the x-lite is failing to register. This should be the case since I don't have any sip entry for 40. Here's the weird part. If I dial 77 from the x-lite phone I get sent to voice mail. If I dial 88 from the x-lite phone I get the demo-congrats message. Why am I getting anything? Why aren't these calls failing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queues, Call groups
Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk no card
You need a timing source for conferencing or music on hold. Voice mail works fine without a timer. If there is no Zaptel card installed, you will have to find timing from a USB driver, or recompile the real time clock. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training Consulting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: Thursday, April 22, 2004 3:57 AM To: asterisk Subject: [Asterisk-Users] asterisk no card Good day all Is it possible to run asterisk and sip without any cards,(t100,voicetronix) Just a plain linux server,running mail and web, and add asterisk At the moment they are running msn? Tanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is anyone successfully using Queues and ACD?
I would like to use queues for auotomated call distribution for a technical support center. Everything I found in the mail archives seemed pretty old. The wikki is pretty sparse. Is anyone successfully using queues for ACD? If so, do you have any examples of /etc/asterisk/*.conf? Thanks! Paul Paul Mahler [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does voice mail require a timer like music on hold and conferencing?
Thanks! Paul Mahler [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP! - weird 7960 problem - phone goes nuts - display flashes - phone reboots
I have a strange problem rolling through my 7960 phones. One or more of the phones goes crazy when the first digit is dialed. The display flashes repeatedly, the phone does a bunch of stuff, sometimes it even reboots. It's not the powered switch, the same thing happens with a different unpowered switch. It's not the phone, the problem moves from phone to phone. If it's happening to a phone and I restart everything, when everything is back up a different phone will have the problem. has anyone seen this? Thanks! Paul Paul Mahler [EMAIL PROTECTED]
RE: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group
Where and when is the rollout meeting? I'd love to attend. Thanks! Paul Paul Mahler [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol Sent: Monday, March 29, 2004 11:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group The VON show has started off with a number of interesting announcements. First among these is a big announcement from Pingtel that they have created a not-for-profit corporation called SIPFoundry. This new company includes Pingtel (which has recently open sourced their SIPExchange PBX), Vovida and somebody else. Martin indicated in his presentation that the key goal of the new group is to leverage the open source SIP implementations to prevent legacy vendors (read Nortel, Avaya, Siemens, etc.) from using the Embrace and Extend model to co-opt and proprietize SIP. Pingtel (which makes SIP hardware) wants to keep the SIP market open and interoperable. They have a web site (which I can't seem to reach from the wireless network here at the show) for the new company/project: http://www.sipfoundry.org I spoke with Martin _ who gave the Pingtel presentation and is an officer of both Pingtel and the new SipForge organization. He indicated he would like to speak with Mark regarding the possibility of integrating the Asterisk community with the SIP Forge community. He indicated that Asterisk was not initially brought into the discussion only due to limited time/resources (and to a lesser degree because Asterisk is not SIP-centric). Can somebody out there take a look at the SIP Forge site and let us all know what the crux of the organization is set to be? They are having an open roll-out meeting tomorrow evening which should spell out some of the goals of the organization and the partners. Above all Martin wanted me to understand that he did not view the new open source organization as a competitor to Asterisk. What do you think? More on-the-scene reports to come. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 SIP Images
I have recieved far more that my money's worth in technical calls to Cisco about my 7960 telephones. They respond immediately. They keep working until the job is done. The pull in whatever resources are neccessary. They have never failed to find and fix the problem. If you want professional, real technical support you should be willing to pay for it, or in this case part of it. Paul Mahler mailto:[EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 27, 2004 7:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 SIP Images What you and so may others on this lise seem to forget is that Cisco is a company offering bsuiness products for businesses. Businesses typically pay by check and wire transfer, especially for items such as this. If you want home-user pay-by-credit-card service, buy products from Belkin's home line and similar. Oh...what's that? None of these cheesy Stocked-at-Costco hardware companies have any VoIP phones worth a crap? Then deal with the fact that you are buying from a company who doesn't target home users, and deal with it. It costs Cisco more money than they make on the contract to offer SmartNet on a single device like this. You're lucky they don't have a minimum device limit/contract cost of something like 5 devices or $300/year. I'm guessing this type of policy would hardly effect more than several hundred of their customers, most of them with 7960's and similar. -Original Message- From: [EMAIL PROTECTED] on behalf of John Baker Sent: Sat 3/27/2004 4:41 PM To: [EMAIL PROTECTED] Cc: Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images [massive amounts trimmed] No, you can't use a credit card. You have to send the #$!@@$#'s a check. It's really stupid, but it's the Cisco way. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: winmail.dat
