Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
I have downloaded iaxmodem and gone through the readme but not yet installed it. I currently use rxfax to receive in the vicinity of 1200 faxes per day and 5000 or more pages (faxes vary from single page to 30 pages) per E1, with a peak load of about 12 concurrent inbound faxes to rxfax. Best I can tell my failure rate is about 0.8%. I have been testing using Hylafax for faxout with an 8 port analog fax modem card and a couple PAP2NA's and this works well, but I am very much looking forward to checking out iaxmodem. Especially if using Hylafax will give me ECM. Craig - Original Message - From: Lee Howard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 13, 2005 10:47 AM Subject: Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable? Darren Nickerson wrote: We prefer the Eicon Diva server and Brooktrout TR1034 boards, which are known to work well with HylaFAX since we've had our share of headaches with the 2977's. Well, part of my preference for the 2977s involves my strong dislike for the way that the Diva Servers and BrookTrouts do things. It's enough of a dislike to get me over the learning curve of how to properly set up the 2977s for HylaFAX use. Having said that, I'm excited to see Lee and Steve improving IAXmodem and the underlying SpanDSP library, and look forward to the day that is performs similarly (or better) to the DSP-laden boards we presently favor! If your favor involves V.34 then it may be a while before the relevant patents expire. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice Outages?
I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week? ___ --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] Broadvoice Outages?
--- Nate Kapi [EMAIL PROTECTED] wrote: I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week? Service is down right now __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.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] displaying a message on the Snom 320 using sipsak
Hi, for the snom360 it is working the same way. Use firmware version 4.3 and be aware that the message is send to a specific SIP line and the phone is displaying it only if this SIP line is the current active line (outgoing identity) symbolised by a black phone (snom360) or the text in brackets (snom320). Regards, Sven Fischer On Wednesday 12 October 2005 20:20, Franklin Webb wrote: Greetings fellow list members, It seems like a lot of people have been having trouble getting indicators working on the Snom phones, myself included. Recently I was able to get the desktop functionality of sipsak to work on my Snom320, and I thought I would share what I could with the list. For those not familiar this will replace the standard display when you are not on a call (normally showing the registered extension) with a text message of your choosing. Our intent is to update this when our agents log into, and out of, queues. This will give a visual indicator for agents and supervisors in our call center as to whether or not the phone is logged in, which is a large concern for us, and probably any call center. For the record I tried this with a Snom360 also and could not get it working. 1. Setup the phone in Asterisk as normal 2. Get and install sipsak. It can be found at http://sipsak.org/ http://sipsak.org/ (can be on any machine on your network, we used a Fedora Core 3 machine for this). 3. In the Snom320 Configuration, under the SIP tab of your extensions line (Line 1 for me) make sure Support Broken Registrar is set to on 4. In the Snom320 Configuration, under Advanced make sure Filter Packets from Registrar is set to off 5. In the Snom320 Configuration, under Advanced under Network identity (port): set it to 5060 (you might be able to use a different port in here and in the sipsak command, but this is what worked for me. 6. Reboot the phone (just to be sure the settings take) Then use the following sipsak command: sipsak -vvv -M -O desktop -B Test Msg -r 5060 -s sip:[EMAIL PROTECTED] where: Test Msg is the message you want displayed. To turn the message off just set it to empty string (). 5060 is the port, you could try another port here if you set your phone to another port under Advanced 6670 is the extension of the phone 192.168.51.251 is the IP of the PHONE, not the Asterisk server. It does not appear that you can use the IP of the Asterisk server. You can get a list of phones with IPs using the Asterisk command sip show peers. Our intent is to build a simple database matching extension to IP and then execute sipsak commands from a script, probably in the manager API, when agents log in and out that will update the phone display accordingly. I hope this is helpful to some of you. Franklin Webb InterMedia Marketing Solutions -- --- See our FAQs at: http://www.snom.com/faq0.html?L=1 Whitepapers at: http://www.snom.com/white_papers.html --- snom technology AG Gradestraße 46 D-12347 Berlin Sven Fischer fax +49 30 39833111 mailto:[EMAIL PROTECTED] http://www.snom.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] Integrated T1
Yes it will support it, you should look up HDLC on the wiki...I went through this a year ago and had a hard time setting it up. It might be easier now though. I would recommend going another route and getting the data brought in seperately with it's own router. You'll also have better redundancy that way. Good luck, Mitchel On 10/12/05, Samy Antoun [EMAIL PROTECTED] wrote: Hi, We have a Data/Voice service supplied through an integrated T1. Does anyone know if Digium T1 card will support the splitting of the Voice and Data? Regards. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.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 ___ --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] Perplexed - IAX trunk == jitterbuffer or no jitterbuffer
I have 4 * servers interconnected with IAX trunks. Three are on a local LAN, one is accessible over a VPN tunnel out of the office. The IAX peer status (iax2 show peers from the CLI) will sometimes show upwards of 300ms. Considering the lag and distance, I am not entirely surprised. Anyway - my question falls towards the jitterbuffer settings in the iax.conf. Should I or should I not? I seem to come across one document that says to do it to only find another document that says this is not the best option for my particular installation. So I am now perplexed. I did updated the MAX_TIMESTAMP_SKEW value in rtp.c to an increased value (found that in one of the bug trackers) and then recompile. But the other settings, let alone to use the jitterbuffer at all, is still a quandary. These are the latest values I am using: jitterbuffer=yes dropcount=2 maxjitterbuffer=200 maxexcessbuffer=40 minexcessbuffer=5 jittershrinkrate=1 I have changed bandwidth and tos to maximize bandwidth and reliability. What I end up with are calls that sound like the far end is in a helicopter. I can only assume that the packets are ending up out of order. Or...? Any help, assistance, guidance, and past experience is GREATLY appreciated! Thanks! Jason ___ --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] supermicro with asterisk and tdm cards
I guess the 2U is not bad... Im going to call supermicro and check what they have. What kind of CPU are you using guys? Seems supermicro has everythiung except the CPU and the HD right? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Kevin Bockman |Sent: Wednesday, October 12, 2005 3:05 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] supermicro with asterisk and tdm cards | |Cory Andrews wrote: | Yeah I should have picked up on that, single PCI Riser in |this one, so | 1 card. I don't know of any 1U solution out there that |would give you | 3 PCI slots to work with, I think you'll have to go to a 2U at least | to achieve this. |I saw the Dell PowerEdge 1850 has 2 PCI-X on separate busses. |That's the only one I've ever seen. | | |Kevin |___ |--Bandwidth and Colocation sponsored by Easynews.com -- | |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris and Asterisk
trixter http://www.0xdecafbad.com wrote: On Wed, 2005-10-12 at 17:46 -0700, Paul Mahler wrote: 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 How was that determined? I have a problem with a plain number like that, which may have been taken into account, why I am asking... Different cpus operate differently, taking more or less time to complete certain functions. Instruction optimization can go a long way if those instructions are used (not terribly likely if its just pushing bits but there are some for just that). Additionally there is no codec processing (presumably) with TDMoE, does the 30MHz take into account any codec processing or is it literally 30MHz (on what cpu class?!) for just pushing bits? There are other factors, but you did say 'about' so they are optional to this conversation, ie other IRQs on the box, potential for device polling, etc. A tuned system for that specific task (pushing bits between a TDM card and ethernet via TDMoE) may be able to operate at a lower clock speed per channel, but that isnt as important for the initial questions. ___ --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 MHZ is not a valid way of gauging performance. It's all about the MIPS (Millions of Instrictions Per Second), Baby :). I was testing with some of the Soekris boards about a year ago for an client, the need was to make a TDMoE - TDMoE router for a wireless network. (Yes I know that that is a stupid idea, and I told the client that it was a waist of his money to have me try.) the board I was using I think was the 4801, not sure thoe (It was a year ago) but it would pust 48 TDMoE channels at once over 100BaseT ok. So I would think that It would. I was using a customized linux distro, (as in one I created) contact me off list if you would like a copy of the distro. -- Christopher Dobbs Wireless Administrator Valario Inovations ___ --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] Which asterisk-friendly cards are fax-capable?
On Thu, 2005-10-13 at 14:03 +0800, Craig Guy wrote: I have downloaded iaxmodem and gone through the readme but not yet installed it. I currently use rxfax to receive in the vicinity of 1200 faxes per day and 5000 or more pages (faxes vary from single page to 30 pages) per E1, with a peak load of about 12 concurrent inbound faxes to rxfax. Best I can tell my failure rate is about 0.8%. I have been testing using Hylafax for faxout with an 8 port analog fax modem card and a couple PAP2NA's and this works well, but I am very much looking forward to checking out iaxmodem. Especially if using Hylafax will give me ECM. Craig You may have already planned this, but I would be interested in hearing how it works for you. Granted that will take some time for you to even know how well it works ... As a side note I am looking at iaxmodem now (although I am easily distracted) with the hopes of using some of the modem codecs spandsp supports to at least get tdd support working for asterisk, and the end hope of more modem protocols. -- 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 signature.asc Description: This is a digitally signed message part ___ --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] Wanting to Make a PocketPC have a secureConnection to asterisk server
VPN? IAX and an SSH Tunnel? - Original Message - From: Kellner, Peter To: asterisk-users@lists.digium.com Sent: Thursday, October 13, 2005 5:18 AM Subject: [Asterisk-Users] Wanting to Make a PocketPC have a secureConnection to asterisk server Does anyone know of a good solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk server? Thanks Peter Kellner http://PeterKellner.net ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Application: Broadcast
What excatly does it do? What messages does it send out? And what software needs to be configured to listen for these messages? Answer these questions and maybe more people will download the source :-) Steve (Not being an arse just reckon a better description is needed) - Original Message - From: Begumisa Gerald M [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, October 13, 2005 3:08 AM Subject: [Asterisk-Users] New Application: Broadcast Hello, I've released an Asterisk application under the terms of the GNU GPL. You may find it here: http://psg.com/~begg/projects/ A short exerpt from the README: -- Broadcast is an Asterisk (http://www.asterisk.org) application which you may use to send a generic message over TCP/IP to any number of computers running software configured to listen for these types of messages. Being written in C, Broadcast will be dynamically loaded onto the Asterisk program on startup, making it a highly reliable and scalable option when compared with other solutions based on the Asterisk Gateway Interface (AGI) system... -- Hope someone finds it useful! Cheers, Gerald. PS: Sorry for the cross posts! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ACD/queues question
Hello Pedro, you should do this using agent priority groups; this way first all low priority agents are filled, then another group is used up. Thanks l. On Wed, 12 Oct 2005 19:30:43 +0200, Pedro Nunes [EMAIL PROTECTED] wrote: Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk?? Thanks in advance Pedro Nunes -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] No incoming calls from chan_capi 0.6
Armin Schindler wrote: On Sat, 8 Oct 2005, Cedric Fontaine wrote: [logfiles] console = notice,warning,error I don't mean the capi.conf. Do you have an extension in context 'entrant' that matches 9100 ? So I added it in the logger.conf and you were right... There was a problem with matching 9100... So it works now ! Cedric ___ --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] Email to FAX
Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Thanks, Bob. ___ --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] migrated to new ver on voip connection vs1server voicemail problems
Asterisk wasn't correctly identifying that the file is actually wave49. We logged into your server and fixed it. -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] From: Tom Vile [mailto:[EMAIL PROTECTED] Sent: Tuesday, October 11, 2005 10:04 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] migrated to new ver on voip connection vs1server voicemail problems Either Permissions on the directory are incorrect or you have no unavail.wav file. On 10/11/05, Andy Goss [EMAIL PROTECTED] wrote: I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail.Anythoughts?The files are there, so I don't get it.Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wavfile 49Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to openfd on /var/spool/asterisk/voicemail/default/5933/unavail.wavOct 11 19:57:26 WARNING[6587]: file.c:804 ast_streamfile: Unable to open/var/spool/asterisk/voicemail/default/5933/unavail (format ulaw): No such file or di--H. Andy GossNetwork EngineerNetwork Advocates Inc.Main: 502.412.1050DID: 502.992.5933Mobile: 502.387.8216[EMAIL PROTECTED]___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --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] Wanting to Make a PocketPC have a secureConnection to asterisk server
On Thu, 2005-10-13 at 08:16 +0100, Steve Daniels wrote: VPN? IAX and an SSH Tunnel? Does anyone know of a good solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk server? Pocket pc supports VPNs natively. No additional software required, assuming you have something on the server that can talk to it. What that is specifically I dont know but perhaps google can tell you what vpn solutions work with the pocket pc. Its not going to be totally secure, with crypto the questions to answer is 'secure from whom and for how long'. Odds are it will be secure enough for the types of data you would have and the types of people that would likely be in a position to eavesdrop. -- 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 signature.asc Description: This is a digitally signed message part ___ --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] Reset IP PHONE 106
I have lost the password of the telephone, so I must do a reset of the telephone. How can I do? I have a VOISMART telehone: IP PHONE 106 Thanks ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --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] AGI and set_callerid for number and name
What version are you using? Try SetCIDName(Fred) Check voip-info's wiki HTH Steve - Original Message - From: Serge Lhermitte [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, October 12, 2005 5:57 PM Subject: [Asterisk-Users] AGI and set_callerid for number and name Hi, I've been trying to use the set_callerid function in the AGI. It sets the CallerIDname properly but I can't figure out how to set the CallerIDnumber. Is it at at possible ? Cheers. SL ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] parameters documentation
Leaving apart people like Mr Totaro, always speeking about other men without knowing them ( I am a consultant, I am in the Networking since early '90, I never ask anything about NAT..) anyway I am going to put that book in my basket (i have a subscription with Oreilly, I can access a certain number of books for at least one month, II use it especially developing J2EE application.) So thank you for the link Andrea FELIX E SKOWRONEK [EMAIL PROTECTED] To asterisk-users@lists.digium.com Sent by: cc asterisk-users-bo [EMAIL PROTECTED] Subject m.com Re: [Asterisk-Users] parameters documentation 12/10/2005 21.34 Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com The people who have been documenting Asterisk have been working on a book for the last few months, it has been published by O'reilly (Asterisk-The Future of Telephony)and is just now finding it's way into the major bookstores, listed under Open-Source at BarnsNoble. While it will not answer everything asterisk can do, but its glossery and and appendix are very helpful for quick reference. If you have been following the Asterisk Documentation Project, some of it will be old hat, but I'm looking forward to replacing my huge stack of printouts with it. It gives a pretty good overview of VOIP, Networking, Telephony, etc. http://www.oreilly.com/catalog/asterisk/ From: Steve Totaro [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] parameters documentation Date: Wed, 12 Oct 2005 11:17:01 -0400 There is plenty of documentation online for both the 3com and *. You have to have good search skills I guess. 3com has the best knowledge base I have seen. http://knowledgebase.3com.com/ and there are tons of 3com dealers that can help. I think you may need to learn some basic networking before learning asterisk. NAT is a very basic concept in networking as well as ports such as 5060 (standard port for SIP). There is a very steep learning curve for asterisk and networking in general. If you want to learn it then you need to dig into the wiki and read all the posts that come across the user's list (well maybe not all of them). There are plenty of consultants that you can hire if you are not up to it. - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Wednesday, October 12, 2005 10:34 AM Subject: Re: [Asterisk-Users] parameters documentation I come from a NBX100 No documentation available. 1 day it starts saying: syslog full and voicemail stop working No one was able to tell me what was the meaning of that alert . 3COM NBX anyway is a good product, but the price is too high, especially 4 years ago, and especially the price of the telephone is very high. Andrea asterisk [EMAIL PROTECTED] echnologies.com To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 13/10/2005 16.13 Re: [Asterisk-Users] parameters
Re: [Asterisk-Users] ACD/queues question
On Thu, 13 Oct 2005 00:39:06 +0200, Tom Rymes [EMAIL PROTECTED] wrote: What we have done is to set up a single queue that all calls come into. For the agents that we want to be our Front Line (i.e.: Customer Service Reps), we give them a penalty of 0. Our Overflow group (i.e.: Customer service reps who are also dealing with walk-in customers and therefore should not be bothered unless we're really busy) gets a penalty of 1, and our Last Resort (i.e.: Everyone else) people get a penalty of 2. That way, all of the calls are answered by our front line people, unless they are all busy/unavailable. Then, and only then, the calls start going to our overflow people, and if they are also all unavailable, the calls go to our last resort people. Seeing as how we have more than 23 people between the three groups, there should technically be no waiting on hold in the queue, even with the PRI saturated. I don't know if this is what you are looking for, but it works extremely well for us. To whomever coded this feature, THANK YOU! As QM supports per service group call flow analysis, I have helped a number of call centers worldwide in setting up this feature together with the adoption of QM and I can say everybody was quite satisfied with it, as much as you can put up with the added problems of running the Agents module. l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] New Application: Broadcast
On Thu, 2005-10-13 at 08:20 +0100, Steve Daniels wrote: What excatly does it do? What messages does it send out? And what software needs to be configured to listen for these messages? Answer these questions and maybe more people will download the source :-) As was explained to me via private email you would do something like: exten = s,1,answer exten = s,2,broadcast(some arbitrary message here) exten = s,3,blah any of the configured systems would get the message, so if you wanted to broadcast caller id or anything else from within a dialplan you could. Any arbitrary message can be embedded in any dialplan wherever you want. As for the listening that wasnt asked by me nor answered (hard to answer a question that is never asked :) so I cant say. -- 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 signature.asc Description: This is a digitally signed message part ___ --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] Maximum retries exceeded on call.
Using SIP? IAX? One way sound is usually a SIP and nat/firewall problem, make sure ports are forwarded. Steve - Original Message - From: Peter Ankerstål [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, October 12, 2005 10:39 PM Subject: [Asterisk-Users] Maximum retries exceeded on call. I have set up a asterisk-server behind NAT and peers to another asterisk and uses that one for outgoing calls. I have som clients on my asterisk and they could register to it well over internet. Not a problem. But when they try to call me the asterisk-server tells me this: Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 32458501 (Non-critical Response) Configs can be found at http://www.pulia.nu/~peter/asterisk/ When they call me they can hear me but I get no sound. Weird. Any Ideas? -- MVH Peter Ankerstål. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Sangoma AA Series?
Seth Remington wrote: Hello All, I saw an add in my latest Linux Journal advertising Sangoma's new AA series of FXO/FXS analog cards with on-board echo cancellation, but I can't find any information at all on them. Even the link given in the advertisement is a dead end as far as I can tell. Anybody else seen/heard anything about this? Hi, Telephonyware ran the ad, and we weren't quite expecting it to be in everyone's hands so soon -- this was our first Linux Journal ad, so we weren't quite prepared with information on our website. As a stop-gap, we've set up a page with some brief information and a mailing list for more information and to announce pre-orders. This is at http://www.telephonyware.com/sangoma We expect to be able to start accepting pre-orders within the next couple of weeks. Keep an eye on the -biz list for more information about that. In the mean time, here is some more information so this thread hasn't been a waste of time. The new cards will be available soon, and will also have an option for an addon 16 port hardware echo canceller with a 128ms echo tail -- this will be available in early December. The analog boards will consist of three components, a shark board, which is the base PCI board, a daughterboard, and FXO/FXS modules (with two FXO, or two FXS interfaces per module). The first board you have in your system will have the base board, the daughterboard, and one or two FXO or FXS modules. When you want to add more than 4 ports, you add another daughterboard, and one or two more FXO/FXS modules. These basically sit over a PCI slot, and screw into the back of the chassis, but do not actually sit in the PCI slot itself. Everything is then connected via an external backplane, in much the same way a series of SCSI drives are plugged into a daisy chain. Here is a picture of a mainboard, plus a second daughterboard, with the backplane connector all hooked up: http://www.telephonyware.com/images/sangoma_backplane.jpg The number of FXO or FXS ports in a single chassis is limited only by the available slots at the back of your chassis. These boards will be in beta within the next two weeks, and they will be around the same price point as the TDM04B. Regards, Mark Lipscombe -- Telephonyware - Intelligent Telephony Solutions 1-866-864-2304 | 1-408-331-1971 | Ext 6006 http://www.telephonyware.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] Wanting to Make a PocketPC have a secureConnection to asterisk server
What kind of connection? Voice or Configuration? SIP? IAX? ssh? I have an ssh client for PPC in one of my archives somewhere. Sorry, dont know what its called, where its from, its been a while since Ive needed it. But it does exist. If thats what you need, Ill take a look for it on the weekend. Kevin From: Kellner, Peter [mailto:[EMAIL PROTECTED] Sent: October 12, 2005 11:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Wanting to Make a PocketPC have a secureConnection to asterisk server Does anyone know of a good solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk server? Thanks Peter Kellner http://PeterKellner.net ___ --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] Email to FAX
Hi Bob, I've justed looked at inter7 solution and perhaps that is what you're looking for (http://www.inter7.com/?page=astfax) Greetings Otto Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI and set_callerid for number and name
Thanks a lot.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro
Nunes
Sent: mercredi 12 octobre 2005 19:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AGI and set_callerid for number and name
Curse,
Look at this php script ...
Contactlookup.agi
#!/usr/local/bin/php -q
?php
ob_implicit_flush(true);
set_time_limit(6);
$in = fopen(php://stdin,r);
$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
// toggle debugging output (more verbose)
$debug = true;
// Do function definitions before we start the main loop
function read() {
global $in, $debug, $stdlog;
$input = str_replace(\n, , fgets($in, 4096));
if ($debug) fputs($stdlog, read: $input\n);
return $input;
}
function errlog($line) {
global $err;
echo VERBOSE \$line\\n;
}
function write($line) {
global $debug, $stdlog;
if ($debug) fputs($stdlog, write: $line\n);
echo $line.\n;
}
// parse agi headers into array
while ($env=read()) {
$s = split(: ,$env);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($env == ) || ($env == \n)) {
break;
}
}
// main program
echo VERBOSE \Here we go!\ 2\n;
read();
$session = mssql_connect('mssql server' , 'username' , 'password' );
$result = mssql_query(select * from ContactDB WHERE
extension=.$agi['callerid'],$session );
$row = mssql_fetch_array($result);
mssql_close($session);
if ($row['Name'] == ){
write('SET VARIABLE NAME Not Found');
read();
} else {
write('SET VARIABLE NAME '.$row['Name'].'');
read();
}
fclose($in);
fclose($stdlog);
And in extensions.conf
[extensions]
exten = 4501,1,agi,contactlookup.agi
exten = 4501,2,SetCIDName(${Name})
exten = 4501,3,Dial(SIP/421,15)
It looks to an mssql DB, try to find the callerID number in table
extensions, and then sets a variable named Name to the value of
table Name. Cool hah...
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Serge
Lhermitte
Sent: quarta-feira, 12 de Outubro de 2005 17:57
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AGI and set_callerid for number and name
Hi,
I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber.
Is it at at possible ?
Cheers.
SL
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Re: [Asterisk-Users] Email to FAX
On Thu, 2005-10-13 at 09:18 +0200, Bohuslav Coufal wrote: Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Effectively T.37 does that, however what you prolly wanna look at is hylafax to process the emails (perhaps by procmail). From within asterisk how the call gets placed doesnt matter a whole lot, and you do have options but basically what you need is a modem (physical of soft like iaxmodem) and a phone line to transmit (analog or digital bri/e1/t1/j1/etc). If you want the call to go through asterisk you may need slightly more, some way to inject the call into asterisk (FXS port, iaxmodem, whatever, all depending on how you configure the device to receive the faxes by email). You can also outsource, some random sites are listed at http://www.voip-info.org/wiki/view/Asterisk+Email+to+Faxl -- 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 signature.asc Description: This is a digitally signed message part ___ --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] Which asterisk-friendly cards are fax-capable?
I'm trying to figure out what an appropriate deployment model might be. Whether to have iaxmodem installed on the hylafax server with a switched ethernet connection for iax2 to the * server with the PRI, or to have the iaxmodem on the PRI * server and channel the tty comms across the network. I suspect that the latter might work ok over a WAN so I could have a central hylafax server with distributed * servers running iaxmodem at the far end of wan links (up to 100ms latency). The only issue is that I want to retain rxfax on the PRI * servers for incoming faxes. Lee, if I install iaxmodem on a * server for outbound faxing from hylafax, can I still use rxfax on the same server to receive faxes? Craig - Original Message - From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 13, 2005 3:06 PM Subject: Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Email to FAX
Thanks, I'll try it. Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Email to FAX Hi Bob, I've justed looked at inter7 solution and perhaps that is what you're looking for (http://www.inter7.com/?page=astfax) Greetings Otto Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor DTMF problems
On 10/12/05, Mir [EMAIL PROTECTED] wrote: We have discovered a problem with DTMF on Asterisk.We have a setup with a T1 from PSTN going into an Asterisk box, and then out again on T1 and into a normal PBX (EADS)We use it to record all calls going to/from the PBX.The problem is that when we record the calls (with MONITOR command),DTMF tones gets obscured, and is not understood in the other end, if we dont Monitor, there are no problems.It sounds like the tones are cut into two, h hard to explain ...Does this ring a bell at anyone ? We have the exact same problem here.We are also using Asterisk to record the calls. Morten Isaksenhttp://www.misak.dk/blog/ ___ --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] ACD/queues question
Thanks, That will fix my problem... And agent skills, is that possible too?? Thanks again Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lorenzo Emilitri Sent: quinta-feira, 13 de Outubro de 2005 8:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ACD/queues question Hello Pedro, you should do this using agent priority groups; this way first all low priority agents are filled, then another group is used up. Thanks l. On Wed, 12 Oct 2005 19:30:43 +0200, Pedro Nunes [EMAIL PROTECTED] wrote: Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk?? Thanks in advance Pedro Nunes -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD/queues question
Thanks, That will fix my problem... And agent skills, is that possible too?? Thanks again Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: quarta-feira, 12 de Outubro de 2005 23:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ACD/queues question On Oct 12, 2005, at 1:30 PM, Pedro Nunes wrote: Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk?? Thanks in advance Pedro Nunes What we have done is to set up a single queue that all calls come into. For the agents that we want to be our Front Line (i.e.: Customer Service Reps), we give them a penalty of 0. Our Overflow group (i.e.: Customer service reps who are also dealing with walk-in customers and therefore should not be bothered unless we're really busy) gets a penalty of 1, and our Last Resort (i.e.: Everyone else) people get a penalty of 2. That way, all of the calls are answered by our front line people, unless they are all busy/unavailable. Then, and only then, the calls start going to our overflow people, and if they are also all unavailable, the calls go to our last resort people. Seeing as how we have more than 23 people between the three groups, there should technically be no waiting on hold in the queue, even with the PRI saturated. I don't know if this is what you are looking for, but it works extremely well for us. To whomever coded this feature, THANK YOU! To set this up, just edit the queues.conf file and add the penalty to each agent's member = line like this: ; Front-line - Penalty of 0 member = 100,0 ; Overflow - Penalty of 1 member = 101,1 ;Last Resort - Penalty of 2 member = 102,2 Hope that proves useful to someone 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 ___ --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] Which asterisk-friendly cards are fax-capable?
On Thu, 2005-10-13 at 15:55 +0800, Craig Guy wrote: I'm trying to figure out what an appropriate deployment model might be. Whether to have iaxmodem installed on the hylafax server with a switched ethernet connection for iax2 to the * server with the PRI, or to have the iaxmodem on the PRI * server and channel the tty comms across the network. I suspect that the latter might work ok over a WAN so I could have a central hylafax server with distributed * servers running iaxmodem at the far end of wan links (up to 100ms latency). The only issue is that I want to retain rxfax on the PRI * servers for incoming faxes. Based on the docs in iaxmodem its better to have iaxmodem on your asterisk server and hylafax (if needed) on a remote server. The lag issues between iaxmodem and asterisk are more critical than hylafax and iaxmodem. Lee, if I install iaxmodem on a * server for outbound faxing from hylafax, can I still use rxfax on the same server to receive faxes? IAXModem works like an iax client, if you redirect calls to that extension they goto iaxmodem if you dont they are handled elsewhere. Treat that as just another extension for all intents and purposes. Problems however may arise if asterisk is told to redirect all calls with a fax tone to rxfax, so you have to deal with that in your dialplan. You would have to either get clever with the extension or do did based routing ... exten = fax,1,gotoif(something?2:3) exten = fax,2,rxfax(somefile) exten = fax,3,Dial(IAX2/iaxmodemExt,60,R) Although this isnt an issue if you do did based routing and the given did is one or the other for that context but not both. Hope this helps (and I hope I am right, but I have been reading a lot and think I am, I am sure lee will point out anything I got wrong) -- 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 signature.asc Description: This is a digitally signed message part ___ --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] ACD/queues question
Just remember to set your phone in the group with the highest possible priority :) On Thu, 2005-10-13 at 09:36 +0100, Pedro Nunes wrote: Thanks, That will fix my problem... And agent skills, is that possible too?? Thanks again Pedro Nunes -- 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 signature.asc Description: This is a digitally signed message part ___ --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] Email to FAX
Hi, when I try to send fax by example in README I got nothing. On asterisk console i saw this: -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) Channel Zap/4-1 was answered. Launching txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) on Zap/4-1 -- Hungup 'Zap/4-1' but nothing is dialed. Have you any suggestions what I made wrong. I using asterisk 1.2.0beta and receiving faxes and incoming calls works good. Thanks, Bob. Dne čtvrtek 13 říjen 2005 09:34 [EMAIL PROTECTED] napsal(a): Hi Bob, I've justed looked at inter7 solution and perhaps that is what you're looking for (http://www.inter7.com/?page=astfax) Greetings Otto Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P off-hook detection problem
Hi list, I have a Wildcard TDM400P REV I Board 1 with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1' -- Hungup 'Zap/13-1' on my CLI, but with port 3 and 4, I don't see anything. I have tried with the same phone that works well in port 1 and 2, so it's not related to the phone. The configuration for port 3 and 4 is idential to 1 and 2. zap show channel xx does not show anything special and what it show is identical between port 1,2 and 3,4. It's a production system, so it's not easy to stop and start troubleshooting it, certainly not easy to open and swap modules Anybody seen something similar ? Thank Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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] Email to FAX
On Thu, 2005-10-13 at 10:45 +0200, Coufal Bohuslav wrote: Hi, when I try to send fax by example in README I got nothing. On asterisk console i saw this: -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) Channel Zap/4-1 was answered. Launching txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) on Zap/4-1 -- Hungup 'Zap/4-1' http://soft-switch.org/installing-spandsp.html When sending a fax it is more likely you will be calling out to the remote FAX machine. In this case, make your Asterisk call the far FAX machine, and when it answers do: exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller) The addition of |caller will make txfax act as a calling machine, rather than an answering machine. This seems ti imply that txfax() doesnt actually dial anything, you have to do that elsewhere, I suggest you use the outgoing spool directory and place (mv not cp) a file in there. Channel: Zap/1/5551212 Maxretries: 0 Waittime: 20 Application: txfax Data: /tmp/fax.tiff|caller This will cause asterisk to call on Zap/1 and dial the number 5551212, when that answers it will call txfax and pass it the path to the fax file and caller (so it acts like a caller not a server/answering endpoint). -- 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 signature.asc Description: This is a digitally signed message part ___ --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] AGI Variable problem
Hello all,
I try to use a agi script to get a variable from * und put them into a
script which gives me another variablke and put this in *.
My problem is now it seems the var ID is empty coz i always jump into
the result 0 loop.
The $MSN should be in the SetCIDNum.
#!/usr/bin/php -q
?php
include(/var/lib/asterisk/agi-bin/phpagi.php);
$agi = new AGI();
$ID = $agi-get_variable(SIPUSER);
if ($ID['result'] == 0) {
$agi-verbose(SIPUSER not set -- nothing to do);
exit(1);
}
$agi-set_variable(MSN, exec(/var/lib/asterisk/agi-bin/msn4sip 111
222 333 .$ID['data']));
?
Output from asterisk:
-- Executing SetVar(SIP/31-79e2, SIPUSER=31) in new stack
-- Executing AGI(SIP/31-79e2, msn4sip.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/msn4sip.agi
msn4sip.agi: Arrayn(n[code] = 510n[result] = n[data] =
Invalid or unknown commandn)n
msn4sip.agi: SIPUSER not set -- nothing to do
-- AGI Script msn4sip.agi completed, returning 0
-- Executing SetLanguage(SIP/31-79e2, de) in new stack
-- Executing SetCIDNum(SIP/31-79e2, ) in new stack
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Re: [Asterisk-Users] Maximum retries exceeded on call.
Yes, using sip. The ports are forwarded. The calls going to the other asterisk server works fine. The problem occurs only when people who are registred to my server tries to call. On Thu, 13 Oct 2005 08:30:17 +0100 Steve Daniels [EMAIL PROTECTED] wrote: Using SIP? IAX? One way sound is usually a SIP and nat/firewall problem, make sure ports are forwarded. Steve - Original Message - From: Peter Ankerstål [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, October 12, 2005 10:39 PM Subject: [Asterisk-Users] Maximum retries exceeded on call. I have set up a asterisk-server behind NAT and peers to another asterisk and uses that one for outgoing calls. I have som clients on my asterisk and they could register to it well over internet. Not a problem. But when they try to call me the asterisk-server tells me this: Oct 12 23:21:38 WARNING[23360]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 32458501 (Non-critical Response) Configs can be found at http://www.pulia.nu/~peter/asterisk/ When they call me they can hear me but I get no sound. Weird. Any Ideas? -- MVH Peter Ankerstål. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- MVH Peter Ankerstål. ___ --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] Email to FAX
But it seems that Asterisk understand that he has to dial (the dialed number is correct), -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) it seems that zap channel had answered (but nothing to try dial), Channel Zap/4-1 was answered. and lunching txfax Launching txfax(/tmp/ast_fax-1129191936.10240.1804289383.0|caller) on Zap/4-1 and immediately hungup -- Hungup 'Zap/4-1' May be something wrong in zapata.conf? ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] ; language=us signalling=fxs_ks context=default ;context=fax channel = 3-4 Thank for any other sugestions, Bob. Dne čtvrtek 13 říjen 2005 11:18 trixter http://www.0xdecafbad.com napsal(a): On Thu, 2005-10-13 at 10:45 +0200, Coufal Bohuslav wrote: Hi, when I try to send fax by example in README I got nothing. On asterisk console i saw this: -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) Channel Zap/4-1 was answered. Launching txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) on Zap/4-1 -- Hungup 'Zap/4-1' http://soft-switch.org/installing-spandsp.html When sending a fax it is more likely you will be calling out to the remote FAX machine. In this case, make your Asterisk call the far FAX machine, and when it answers do: exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller) The addition of |caller will make txfax act as a calling machine, rather than an answering machine. This seems ti imply that txfax() doesnt actually dial anything, you have to do that elsewhere, I suggest you use the outgoing spool directory and place (mv not cp) a file in there. Channel: Zap/1/5551212 Maxretries: 0 Waittime: 20 Application: txfax Data: /tmp/fax.tiff|caller This will cause asterisk to call on Zap/1 and dial the number 5551212, when that answers it will call txfax and pass it the path to the fax file and caller (so it acts like a caller not a server/answering endpoint). ___ --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] Patton SmartNode
Actually we 're running the sip protocol but in the past we did also use h323 in combination with tedas phoneware server (german voip solution). Both ran on SmartNode side very stable. Caller ID Name with sip/h323 should not be a problem, but here in Germany I'm not really shure, if the telco (T-COM) does support this feature on the PSTN side. I guess the SmartNodes should do the job anyway. Regards Guido Hecken Are you running SIP, or H323, or MGCP? Also, do you get callerid name passed through? Guido Hecken wrote: We use the SmartNodes SN1400 and SN2300 as ISDN Gateways in our customer Asterisk installations and are really happy with them. They run very stable and you can configure nearly everything. Support from INALP is also great. With the interface cards for the SmartNode 2300 you should be able to connect nearly everything to VOIP. Regards Guido Hecken Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible alternative. Thanks. Peder ___ --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] Email to FAX
Yeah I missed that in the original, sorry bout that. are you sure that the other end didnt hang up? You may want to test this by calling a number you have access to so that you can at least rule that out. The only other thing I can think of is that txfax itself is aborting and returning prematurely. I wonder if its a negotiation failure. You say it hangs up immediatly, how immediatly? 1 second? 5? On Thu, 2005-10-13 at 11:52 +0200, Coufal Bohuslav wrote: But it seems that Asterisk understand that he has to dial (the dialed number is correct), -- Attempting call on Zap/4/585228796 for application txfax(/tmp/ast_fax-1129192102.10240.1804289383.0|caller) (Retry 1) it seems that zap channel had answered (but nothing to try dial), Channel Zap/4-1 was answered. and lunching txfax Launching txfax(/tmp/ast_fax-1129191936.10240.1804289383.0|caller) on Zap/4-1 and immediately hungup -- Hungup 'Zap/4-1' May be something wrong in zapata.conf? ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [channels] ; language=us signalling=fxs_ks context=default ;context=fax channel = 3-4 Thank for any other sugestions, Bob. -- 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 signature.asc Description: This is a digitally signed message part ___ --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] pbx_spool Call failed to go through
Hello Im getting this error any body have any idea how to fix it pbx_spool.c:229 attempt_thread: Call failed to go through, reason 3 Regards Fahd Ansari ___ --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] TDM400P off-hook detection problem
Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. Are you sure that in 3rd and 4th ports you have immediate=no? regards, srsergio -Mensaje original- De: Alex Ongena [mailto:[EMAIL PROTECTED] Enviado el: jueves, 13 de octubre de 2005 11:17 Para: Asterisk Asunto: [Asterisk-Users] TDM400P off-hook detection problem Hi list, I have a Wildcard TDM400P REV I Board 1 with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1' -- Hungup 'Zap/13-1' on my CLI, but with port 3 and 4, I don't see anything. I have tried with the same phone that works well in port 1 and 2, so it's not related to the phone. The configuration for port 3 and 4 is idential to 1 and 2. zap show channel xx does not show anything special and what it show is identical between port 1,2 and 3,4. It's a production system, so it's not easy to stop and start troubleshooting it, certainly not easy to open and swap modules Anybody seen something similar ? Thank Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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 Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.14/131 - Release Date: 12/10/2005 ___ --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] TDM400P off-hook detection problem
On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote: Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. sorry, my mistake, these are all FXS ports with FXO signaling. I always mix them up. Are you sure that in 3rd and 4th ports you have immediate=no? my zapate.conf file: (channel 13 and 14 are ok, 15 and 16 give problems) [channels] ; Default language language=be callerid=asreceived immediate=no switchtype=euroisdn ; ; Poort 1-2 gaan naar de PBX ; context=isdn-pbx signalling=bri_net group=2 channel = 1-2,4-5 echocancel = yes ; ; Poort 3-4 gaan naar Publiek net ; context=isdn-public signalling = bri_cpe group=1 channel = 7-8,10-11 ; ; TDM40B kanalen ; signalling=fxo_ks language=be context=analog echocancel=no channel = 13 signalling=fxo_ks language=be context=analog channel = 14 signalling=fxo_ks language=be context=analog channel = 15 signalling=fxo_ks language=be context=analog channel = 16 Thanks already alex regards, srsergio -Mensaje original- De: Alex Ongena [mailto:[EMAIL PROTECTED] Enviado el: jueves, 13 de octubre de 2005 11:17 Para: Asterisk Asunto: [Asterisk-Users] TDM400P off-hook detection problem Hi list, I have a Wildcard TDM400P REV I Board 1 with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1' -- Hungup 'Zap/13-1' on my CLI, but with port 3 and 4, I don't see anything. I have tried with the same phone that works well in port 1 and 2, so it's not related to the phone. The configuration for port 3 and 4 is idential to 1 and 2. zap show channel xx does not show anything special and what it show is identical between port 1,2 and 3,4. It's a production system, so it's not easy to stop and start troubleshooting it, certainly not easy to open and swap modules Anybody seen something similar ? Thank Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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 Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.14/131 - Release Date: 12/10/2005 ___ --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 -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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] USB phone for Linux?
Hi, Can anyone recommend a USB phone that can be used under Linux, either interfacing directly with Asterisk in some way, or using a soft phone program on Linux that doesn't need screen interaction (only using the phone's keypad)? The idea is to be able to plug it into the USB port of an Asterisk box in a rack, where screen, kbd and mouse may not be available. Thanks in advance! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] TDM400P off-hook detection problem
On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote: Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. Are you sure that in 3rd and 4th ports you have immediate=no? if it may help, I could just stop *, # rmmod wcfxs # modprobe wcfxs # asterisk and now all ports are working fine ??? I Googled around and found someone with a similar problem 5 okt 2004. It happened after 2 weeks of operation ? I think it's still an issue in the driver... Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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] Broadvoice Outages?
Yes, i am having timeouts on registering to the LAX sip server of broadvoice. Marco. Nate Kapi wrote: I've been having a lot of problems with Broadvoice lately. Anyone else been without service for extended periods of time this week? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P off-hook detection problem
Check your Revision card, if it is Rev H in zaptel sources you have a zconfig.h with a Flag to Revision H. Try it. regards, -Mensaje original- De: Alex Ongena [mailto:[EMAIL PROTECTED] Enviado el: jueves, 13 de octubre de 2005 12:56 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] TDM400P off-hook detection problem On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote: Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. Are you sure that in 3rd and 4th ports you have immediate=no? if it may help, I could just stop *, # rmmod wcfxs # modprobe wcfxs # asterisk and now all ports are working fine ??? I Googled around and found someone with a similar problem 5 okt 2004. It happened after 2 weeks of operation ? I think it's still an issue in the driver... Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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 Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.14/131 - Release Date: 12/10/2005 ___ --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
[Asterisk-Users] Music on hold disappears for Dial(, m) when calling outside numbers
My asterisk is purely connected to the outside world via SIP. When I use Dial() with the m-option, that should ensure music-on-hold, it works perfectly as long as I am calling a SIP number, but when I call a mobile phone, the music-on-hold disappears. Any ideas on the cause of this or how to fix this? Lars Dybdahl. ___ --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] IAX ATA
Hi! Has anyone tested this IAX ATA? Their free softphone is GREAT https://www.virbiage.com/products.php Regards Anders Svensson ___ --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] SetCallerID Problem
My number is not submitted. I updated my asterisk but this error still occurs coz of the in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum(SIP/31-752a, 4989427) in new stack -- Executing SetCIDName(SIP/31-752a, 4989427) in new stack -- Executing SetCallerID(SIP/31-752a, 4989427 4989427) in new stack ___ --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] IAX ATA
Has anyone tested this IAX ATA? https://www.virbiage.com/products.php For some reason, their IAX hardphone was coming soon for two years on the site and then... still no word. ___ --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] sangoma a104 cards and ss7 signaling
Hi Sangoma a104 card have in product specyfication support for Line protocol SS7 , http://www.sangoma.com/products/p_aft-104-specs.htm [..] Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. [..] Anyone of you guys use line protocol SS7 for E1/T1 termination in asterisk ? As far I know asterisk don't have support for SS7 signaling, but my telco wants to setup E1 link with SS7 signaling and suggest sangoma a104. /pch -- Dyslexia bug unpatched since 1977 ... exploit has been leaked to the underground. ___ --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] Starting simple switch from an extension?
Hi, Is there a command to start simpleswitch from an extension? For example it would allow me to dial in to my * box and get a dial tone to make an outgoing call. Thanks, Derek -- Derek Conniffe Rivertower Ltd DID Number: 01 440 1806 (International: 00 353 1 440 1806) Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 440 1800 United Kingdom: 0870 068 2368 International: 00 353 1 440 1800 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com begin:vcard fn:Derek Conniffe n:Conniffe;Derek org:Rivertower Ltd;IT adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 201 0146 tel;fax:+353 1 201 0085 tel;cell:+353 86 856 3823 note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A= Ireland: (Local) 01 244 9719=0D=0A= United Kingdom: 0870 068 2368=0D=0A= International: 00 353 1 244 9719=0D=0A= url:http://www.rivertowerhosting.com version:2.1 end:vcard ___ --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] Starting simple switch from an extension?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA is probably what you need. Thanks, Steve Totaro - Original Message - From: Derek Conniffe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 13, 2005 7:58 AM Subject: [Asterisk-Users] Starting simple switch from an extension? Hi, Is there a command to start simpleswitch from an extension? For example it would allow me to dial in to my * box and get a dial tone to make an outgoing call. Thanks, Derek -- Derek Conniffe Rivertower Ltd DID Number: 01 440 1806 (International: 00 353 1 440 1806) Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 440 1800 United Kingdom: 0870 068 2368 International: 00 353 1 440 1800 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.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 No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.14/130 - Release Date: 10/12/05 ___ --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] SetCallerID Problem
René Enskat [Teamware GmbH] wrote: My number is not submitted. I updated my asterisk but this error still occurs coz of the in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum(SIP/31-752a, 4989427) in new stack -- Executing SetCIDName(SIP/31-752a, 4989427) in new stack -- Executing SetCallerID(SIP/31-752a, 4989427 4989427) in new stack I'm having the same issue. Looking forward to a fix. Doug -- Ben Franklin quote: Those who give up essential liberties for temporary safety deserve neither liberty nor safety. ___ --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] Perplexed - IAX trunk == jitterbuffer or no jitterbuffer
On Wed, 12 Oct 2005, Jason Walker wrote: I have 4 * servers interconnected with IAX trunks. Three are on a local LAN, one is accessible over a VPN tunnel out of the office. The IAX peer status (iax2 show peers from the CLI) will sometimes show upwards of 300ms. Considering the lag and distance, I am not entirely surprised. Anyway - my question falls towards the jitterbuffer settings in the iax.conf. Should I or should I not? I seem to come across one document that says to do it to only find another document that says this is not the best option for my particular installation. So I am now perplexed. Hi Jason, You need to tell us which Asterisk version you are using. In the 1.0 series, trunking and the jitter buffer won't work together - the trunking process mangles frame timestamps in a way that the jitter buffer can't handle. In CVS-HEAD/1.2, you can optionally have trunked frames include extra timestamp info so that the jitter buffer can still work. Regards, Steve ___ --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] polycom soundpoint ip600 problem
Hello, I have a polycom ip600 and eyebeam. When I call from polycom to eyeBeam, everything, including audio works. When I call the other side (from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows the same codec: g711u. Also sip show channels shows ulaw codec for both sides and correct addresses. I have canreinvite=no. I don't know if it's important, but asterisk console shows me warning chan_sip.c:3250 process_sdp: Error in codec string 'eo 0 sip 34 103'. Running CVS Head, some older build. Any ideas what could be wrong will be very helpful. Juraj. ___ --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] USB phone for Linux?
Tony Mountifield wrote: Hi, Can anyone recommend a USB phone that can be used under Linux, either interfacing directly with Asterisk in some way, or using a soft phone program on Linux that doesn't need screen interaction (only using the phone's keypad)? The idea is to be able to plug it into the USB port of an Asterisk box in a rack, where screen, kbd and mouse may not be available. Thanks in advance! Tony Find me a USB phone with sufficient hardware docs available and I will see what I can do. I could use the same type of thing. I have remote customer servers and would love to have them setup so my contractor tech can just plug in and become extension on my pbx here. What I would do is base the softphone on something like iaxclient. I would have it launched when the usb hotplug was seen. I suppose this could be initially done with 2 devices. One would be a good usb headset and the other would be a keypad with lcd display. ___ --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] Music on hold disappears for Dial(, m) when calling outside numbers
Try disabling inband call progress tones. Let Asterisk handle everything. In sip.conf add the line: progressinband=no On 10/13/05, Lars Dybdahl [EMAIL PROTECTED] wrote: My asterisk is purely connected to the outside world via SIP. When I use Dial() with the m-option, that should ensure music-on-hold, it works perfectly as long as I am calling a SIP number, but when I call a mobile phone, the music-on-hold disappears. Any ideas on the cause of this or how to fix this? Lars Dybdahl. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PickUpChan and Intercept
Hello everyone,
I have been asked for "directed pickup" and saw
that both "PickupChan" from bristuff and "Intercept" applications
do the dirty work.
I have tried both on asterisk-1.0.9 (
BRIstuffed-0.2.0-RC8o ) but I always got an error when trying to pick the
ringing call.
the debug says:
SIP/marco-73a0 is
ringing -- SIP/marco-73a0 is ringing
-- SIP/marco-73a0 is ringing -- SIP/marco-73a0 is
ringing -- Executing PickupChan("SIP/edevena-a940",
"SIP/marco") in new stackOct 13 12:35:48 WARNING[1675]: channel.c:513
ast_channel_walk_locked: Avoided initial deadlock for 'SIP/edevena-a940',
10 retries! -- No channel found
SIP/marco. -- SIP/marco-73a0 is ringing
the problems seems in ast_channel_walk_locked.
Will someone help on this matter?
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Re: [Asterisk-Users] IAX ATA
Are you looking on purchasing one? francis www.VoIPware.ca On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote: Hi! Has anyone tested this IAX ATA? Their free softphone is GREAT https://www.virbiage.com/products.php Regards Anders Svensson ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards,Francis BallaresE-mail: ballares (at) gmail.com www.VoIPware.ca ___ --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] link quality monitor
hi, do you someone know tool that can get data like latency/bandwith/jitter/packet loss (in one program) - it must be functional behind nat - multiplatform (AJAX,java applet) - preferably on SIP and IAX ports - can be client/server - easy to use ;) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --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
[ SOLVED ] [Asterisk-Users] ISDN problem: lacking dialtone
Title: Patrick Briefpapier Hi Martin, I saw your problem listing on the Asterisk mail archives. I seem to have the same problem with the ISDN 'lacking dialtone' message I still have not been able to get it working, could you share your modem / extension / sip conf files? Thanks in advance! - Patrick --- This email was scanned by MyMail of DatacomPartner http://www.datacompartner.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] Which asterisk-friendly cards are fax-capable?
Craig Guy wrote: I have downloaded iaxmodem and gone through the readme but not yet installed it. I currently use rxfax to receive in the vicinity of 1200 faxes per day and 5000 or more pages (faxes vary from single page to 30 pages) per E1, with a peak load of about 12 concurrent inbound faxes to rxfax. Best I can tell my failure rate is about 0.8%. I have been testing using Hylafax for faxout with an 8 port analog fax modem card and a couple PAP2NA's and this works well, but I am very much looking forward to checking out iaxmodem. Especially if using Hylafax will give me ECM. No. No. This can't be right. We've been hearing authoritative statements on this mailing list that soft modems can't possibly work. :-) I have no clear idea how many people actually use my software for fairly high volumes. There are now clearly many thousands successfully using it for modest levels of faxing. I have heard from a few people doing rather higher volumes than you. Other people have problem - I mean genuine problems, rather than the frame slips issues. I don't get enough feedback to really work out what it going on with those troublesome installations. Regards, Steve ___ --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] PA168S/AT320P
Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --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] Re: Canadian Association of VoIP Providers
John Lange [EMAIL PROTECTED] wrote: My apologies for the cross-posting. If you think you should apologize for it, don't do it. If you think it is okay to do it, don't apologize. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.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] Email to FAX
Also take a look at www.trustfax.com They've done a fine job for us and have several different plans that address from very low to high volume faxing. Receiving faxes via email as pdf files is great, very timely, with no errors identified in the past six months. From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Email to FAX Date: Thu, 13 Oct 2005 02:34:39 -0500 (CDT) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi Bob, I've justed looked at inter7 solution and perhaps that is what you're looking for (http://www.inter7.com/?page=astfax) Greetings Otto Hi all, Does anybody has good working solution for email to fax (simply sending faxes) by asterisk. Thanks, Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --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] PA168S/AT320P
1)What is the protocol you are using? SIP or IAX2? 2)Have you applied the correct firmware to the Phone? Pa168 phones are falwless when connecting to Asterisk. Start the configuration as asimple entry as under. I have added Port address and allowed codecs in the config below: [221] type=friend username=221 secret=secret context=local host=dynamic dtmfmode=rfc2833 nat=yes Port=5060 Disallow=all Allow=g729 Allow=ulaw Allow=gsm Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168S/AT320P Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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] which voip fone will be better
hy all i want to knwo that which voip fone( hard fone ) will be better either it should be iax, sip or h.323 ( that should be good and not too expensive ) i want to have a setup of 200 fones in five offices. and is there any card available to connect four pstn lines. like in single channel fxo there is only one channel. waiting for a good suggestion thankx ishtiaq ahmed __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --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] link quality monitor
Hi, Iperf does it, but is not made for running as MRTG or Nagios. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of marek cervenka Sent: Thursday, October 13, 2005 9:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] link quality monitor hi, do you someone know tool that can get data like latency/bandwith/jitter/packet loss (in one program) - it must be functional behind nat - multiplatform (AJAX,java applet) - preferably on SIP and IAX ports - can be client/server - easy to use ;) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Canadian Association of VoIP Providers
Hi, please add me to the mailing list I also can donate webspace, bandwidth, IAX local dialtone to 780 area code, and DNS services. btw how are you going to do the conference call, with MeetMe? -Original Message- From: John Lange [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 12, 2005 1:17 PM To: Asterisk; Commercial and Business-Oriented Asterisk Discussion; Asterisk Developers Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion; Cisco-VoIP Subject: [Asterisk-Users] Canadian Association of VoIP Providers My apologies for the cross-posting. If you are a business or individual providing Voice over IP services in Canada then we encourage you to read this email carefully otherwise please disregard. - As you are most likely aware, the CRTC has undertaken the roll of regulating VoIP services in Canada and is currently conducting hearings with the goal of putting in place regulatory requirements for all VoIP providers. Specifically, the CRTC's CISC VoIP 911 working group ( http://www.crtc.gc.ca/cisc/eng/cisf3e4_20.htm ) is very actively looking at what regulations to put in place in order to implement E911 services for VoIP. The recommendations of this committee will have direct impact on your business. Currently this working group is is largely comprised by the Local Exchange Carriers (ILECs CLECs) with representation from the large VoIP providers (Primus Vonage). To date only a very few smaller VoIP providers are participating. Subsequently, much of the discussion is oriented around solutions designed to work in the traditional telco world. Depending on your companies infrastructure these solutions may be very expensive or completely impossible for your business to implement. Some members of the working group are even of the position that VoIP service be abolished altogether. Your companies direct participation in the hearings is the best way to have an impact. However, we acknowledge that not all companies have the time and/or resources to fully participate lengthy public hearings. It is with this in mind we propose the formation of a Canadian industry association for VoIP providers and we invite you to participate. The short term goal is to contact and organize Canadian VoIP providers into a formal association. Longer term the association will work towards the following goals: - Keep VoIP providers informed about current regulatory issues - Ensuring VoIP providers have a place at the CRTC table - Develop industry recommendations - Communicate industry recommendations to the CRTC working group - Communicate industry positions to the media - Other (to be determined by the association) At the outset it is envisioned that this group would work in the following way: - No membership fee - Regular updates via email list - Frequent Conference calls - No face-to-face meetings (no travel) - Development of an Industry web site - In-person representation at each CRTC meeting (The CRTC working group meets monthly in a different province each month. We hope to have at least one member representative attend each meeting.) To voice your support (or opposition) for the formation of this group please contact me directly either by email or telephone (contact information in the signature). It is important that you do not delay. CISC working group recommendations to the CRTC are forthcoming. You will be contacted with details on how to participate in the formation of this association. Our intention is to hold our first conference call as early as possible (early next week). NOTE: No web site or association material yet exists because the group has not been officially formed and named. This will be one of the first items of business for the new group. Regards, -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Impport script for upgrading to 1.2 SQL Realtime?
Hi, Is there a script anywhere which would import existing *.conf entries into a mysql database for use with the realtime architecture? Thanks in advance. -- -Barry Flanagan ___ --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] Noob help with IAX
Ok so I've just built and installed a CVS (HEAD) version of asterisk on RHFC2 running a 2.6.13.3 kernel.org kernel. I installed the samples via make samples. Everything seems to work except one thing. I'm trying to do the connect to the Digium IAX demo server portion of the demo (dial 500) and I just get the following messages. I am behind a NAT server and did NOT change anything in any of the sample config files from CVS. Could this be the problem? BTW - I'm using the Xlite soft phone running on the same box as the asterisk server. -- Executing Playback(SIP/xlite1-625c, demo-abouttotry) in new stack -- Playing 'demo-abouttotry' (language 'en') -- Executing Dial(SIP/xlite1-625c, IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED]/[EMAIL PROTECTED] -- IAX2/216.207.245.8:4569-1 is circuit-busy Oct 13 08:45:56 NOTICE[20718]: chan_iax2.c:2754 auto_congest: Auto-congesting call due to slow response -- Hungup 'IAX2/216.207.245.8:4569-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Playback(SIP/xlite1-625c, demo-nogo) in new stack -- Playing 'demo-nogo' (language 'en') == Spawn extension (default, 500, 3) exited non-zero on 'SIP/xlite1-625c' -- Saved useragent X-Lite release 1105d for peer xlite1 -- Michael J. Lynch What if the hokey pokey IS what it's all about -- author unknown ___ --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] IAX ATA
Yes I was interested to test them. They are not available on the link you submitted either Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francis Ballares (VoIPware.ca) Sent: den 13 oktober 2005 15:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX ATA Are you looking on purchasing one? francis www.VoIPware.ca On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote: Hi! Has anyone tested this IAX ATA? Their free softphone is GREAT https://www.virbiage.com/products.php Regards Anders Svensson ___ --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 -- Regards, Francis Ballares E-mail: ballares (at) gmail.com www.VoIPware.ca ___ --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] Mail server question
Hi there: I have a simple question...can I use the internal mail server that uses * as my organization pop-smtp server, if so how can I do it. Thanks Hector ___ --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] Moscow Dids
Hello I need Moscow dids urgently, Contact me offline [EMAIL PROTECTED] Regards Mehdi Chouikh Universal Telecom Spain ___ --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] RE: Wanting to Make a PocketPC have asecureConnection to asterisk server
Im wanting both the voice and the configuration to be secure. (very secure). I dont care if it is SIP or IAX but I do need a softphone on the pocketpc I can use. Id appreciate if you could take a look this weekend for me. Thanks, -Peter From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Scott Sent: Thursday, October 13, 2005 12:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Wanting to Make a PocketPC have asecureConnection to asterisk server What kind of connection? Voice or Configuration? SIP? IAX? ssh? I have an ssh client for PPC in one of my archives somewhere. Sorry, dont know what its called, where its from, its been a while since Ive needed it. But it does exist. If thats what you need, Ill take a look for it on the weekend. Kevin From: Kellner, Peter [mailto:[EMAIL PROTECTED] Sent: October 12, 2005 11:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Wanting to Make a PocketPC have a secureConnection to asterisk server Does anyone know of a good solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk server? Thanks Peter Kellner http://PeterKellner.net ___ --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] AGI Variable problem
for some reason your script is not executing the get_var correctly, as
you can see in the output, asterisk is saying: invalid or unknown
command.
check the internals of your script, the most common reason is that you are mispelling the command.
best regardsOn 10/13/05, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote:
Hello all,I try to use a agi script to get a variable from * und put them into ascript which gives me another variablke and put this in *.My problem is now it seems the var ID is empty coz i always jump into
the result 0 loop.The $MSN should be in the SetCIDNum.#!/usr/bin/php -q?phpinclude(/var/lib/asterisk/agi-bin/phpagi.php);$agi = new AGI();$ID = $agi-get_variable(SIPUSER);
if ($ID['result'] == 0) {$agi-verbose(SIPUSER not set -- nothing to do);exit(1);}$agi-set_variable(MSN, exec(/var/lib/asterisk/agi-bin/msn4sip 111
222 333 .$ID['data']));?Output from asterisk:-- Executing SetVar(SIP/31-79e2, SIPUSER=31) in new stack-- Executing AGI(SIP/31-79e2, msn4sip.agi
) in new stack-- Launched AGI Script /var/lib/asterisk/agi-bin/msn4sip.agimsn4sip.agi:
Arrayn(n[code] =
510n[result] =
n[data] =Invalid or unknown commandn)nmsn4sip.agi: SIPUSER not set -- nothing to do-- AGI Script msn4sip.agi completed, returning 0-- Executing SetLanguage(SIP/31-79e2, de) in new stack
-- Executing SetCIDNum(SIP/31-79e2, ) in new stack___--Bandwidth and Colocation sponsored by Easynews.com
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[Asterisk-Users] CallerID detection problem
hi, is there anyway to make * to detect callerid before first ring. i know that it seems silly; but here i have a case that Telco sends the caller-id before first ring. this issue is detected by installing a callerid detection device on the line. it shows callerid just before the first ring. so * can't detect the callerid. thanks, paradise dove ___ --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] sip register incoming call contexts?
I did try this and did get it to register as this peer. However inbound calls to that number are still coming into the context defined in [general] sip.conf I now have two numbers configured, the new peer as you sugested and my original that just has the register line without an associated peer section. BOTH numbers are still coming into the context defined in [general] THis is fine for the number of which I did not create a peer section for. The other number that I did indeed create a peer section for is not coming into the context that I set within the peer context= I of course am doing a full stop of asterisk and restart/reload for each test. Am I still doing something wrong here? Thanks! Steve Create a peer with a host= setting that matches the IP of the service provider's proxy. Set context for this peer. There are several examples out there, one is http://edvina.net/broadvoice/ /Olle Steve Gladden wrote: Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register = nnn:[EMAIL PROTECTED] -or- register = nnn:[EMAIL PROTECTED]/nnn to come directly into an extension in the dialplan It seems that this only works with the default context in the dialplan. I have another sip account from another provider that I would like all of it's incoming calls to come into the s, extension of a new context but I have been unable to figure out how to bring calls from a register line into an alternate context. Create a peer with a host= setting that matches the IP of the service provider's proxy. Set context for this peer. There are several examples out there, one is http://edvina.net/broadvoice/ /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PA168S/AT320P
Hi, thanks to reply: 1)SIP 2)yes. I've used the original 1.46 for SIP protocol Also your solution do not work. Are 2 days that I'm trying configurations and googling for this problem, but nothing! Always: LOG ON FAILED I've saw about problems with this phone, but my hope was that with the new firmware something could be solved. Thanks again. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: 1)What is the protocol you are using? SIP or IAX2? 2)Have you applied the correct firmware to the Phone? Pa168 phones are falwless when connecting to Asterisk. Start the configuration as asimple entry as under. I have added Port address and allowed codecs in the config below: [221] type=friend username=221 secret=secret context=local host=dynamic dtmfmode=rfc2833 nat=yes Port=5060 Disallow=all Allow=g729 Allow=ulaw Allow=gsm Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168S/AT320P Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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 -- .:FaberK:. ___ --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] fax consult
Dear sirs,
I believe that this question should go to Steve Underwood,
but if someone else also has something to say, I have my ears totally open.
After differents tests (None of them worked), Im
ready to install spandsp, app_txfax app_rx fax to try fax to email
email to fax.
This is my environment.
A server with a PRI card Digium TE405, a Pentium 4 HT 3.2 GHz
with 1 GB ram, RH 9.0 and Asterisk CVS-v1-0-12/28/04-03:08:11 is all that I can
get from show version.
I already installed:
autoconf 2.59
automake 1.9.6
libtool 1.5.20
GNU m4 1.4.3
Libtiff 3.7.4
Jpeg (However Libtiff doesnt find it)
astfax installed
epstools installed
spandsp installed
See the report:
Installation directory: /usr/local
Documentation directory:
${prefix}/share/doc/tiff-3.7.4
C compiler: gcc -g -O2 -Wall
C++ compiler: g++ -g -O2
Enable runtime linker paths: no
Support for internal codecs:
CCITT Group 3 4 algorithms: yes
Macintosh PackBits algorithm: yes
LZW algorithm: yes
ThunderScan 4-bit RLE algorithm: yes
NeXT 2-bit RLE algorithm: yes
LogLuv high dynamic range encoding: yes
Support for external codecs:
ZLIB support: yes
Pixar log-format algorithm: yes
JPEG support: no
Old JPEG support: no
C++ support: yes
OpenGL support: no
And it left to compile app_rx_fax app_tx_fax.
It is any other recommendation before to start with this?
Thanks for any comment,
Carlos Alperin
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Re: [Asterisk-Users] Noob help with IAX
I just tested it and it's working fine. Does your Linux box have internet access? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] TDM400P off-hook detection problem
It's a REV I ... Txs On Thu, 2005-10-13 at 13:06 +0200, Sergio Serrano wrote: Check your Revision card, if it is Rev H in zaptel sources you have a zconfig.h with a Flag to Revision H. Try it. -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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] Noob help with IAX
Matt Riddell wrote: I just tested it and it's working fine. Does your Linux box have internet access? Yep, but through a firewall. I figured it probably works ok and that I must just be doing something wrong. The only config file I changed was sip.conf. In this file I just uncommented out the xlite1 section to make the xlite soft phone work. Like I said, everything else seems to work (E.G. when I dial 1000, I get the successful install message) -- Michael J. Lynch What if the hokey pokey IS what it's all about -- author unknown ___ --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] IAX ATA
I have other IAX ATA's available at VoIPware.ca - I have tested them personally and they work great. thanks, Francis www.VoIPware.ca On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote: Yes I was interested to test them. They are not available on the link you submitted either Anders From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Francis Ballares (VoIPware.ca)Sent: den 13 oktober 2005 15:12 To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] IAX ATA Are you looking on purchasing one? francis www.VoIPware.ca On 10/13/05, Anders Svensson [EMAIL PROTECTED] wrote: Hi! Has anyone tested this IAX ATA? Their free softphone is GREAT https://www.virbiage.com/products.php Regards Anders Svensson ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Francis BallaresE-mail: ballares (at) gmail.comwww.VoIPware.ca ___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards,Francis BallaresE-mail: ballares (at) gmail.comwww.VoIPware.ca ___ --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] fax consulting
I want to modify the info Libtiff is 3.5.7 (uninstalled the 3.7.4 and install this one after reading a note about the crash) Audiofile is 0.2.6 Thanks, Carlos Alperin ___ --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] Starting simple switch from an extension?
DISA(password|context) On Thu, 2005-10-13 at 12:58 +0100, Derek Conniffe wrote: Hi, Is there a command to start simpleswitch from an extension? For example it would allow me to dial in to my * box and get a dial tone to make an outgoing call. Thanks, Derek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PA168S/AT320P
have you configured the STUN server on the phone to any one of the available stun servers like stun.xten.net? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PA168S/AT320P Hi, thanks to reply: 1)SIP 2)yes. I've used the original 1.46 for SIP protocol Also your solution do not work. Are 2 days that I'm trying configurations and googling for this problem, but nothing! Always: LOG ON FAILED I've saw about problems with this phone, but my hope was that with the new firmware something could be solved. Thanks again. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: 1)What is the protocol you are using? SIP or IAX2? 2)Have you applied the correct firmware to the Phone? Pa168 phones are falwless when connecting to Asterisk. Start the configuration as asimple entry as under. I have added Port address and allowed codecs in the config below: [221] type=friend username=221 secret=secret context=local host=dynamic dtmfmode=rfc2833 nat=yes Port=5060 Disallow=all Allow=g729 Allow=ulaw Allow=gsm Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168S/AT320P Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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 -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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] Not ringing on incoming callls
Anyone have any ideas as to why a call coming in won't ring the phone? I can call the phone from my cell and when I hear it ringing on the cell phone I pick up the house phone that should be ringing and am able to talk. I have tried two different pap2-na adapters, have verified the ports on my firewall and also a couple of different house phones. I am not running Asterisk yet but will be once I figure out this problem. Thanks, Travis ___ --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] PA168S/AT320P
Right now, but nothing changed. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: have you configured the STUN server on the phone to any one of the available stun servers like stun.xten.net? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PA168S/AT320P Hi, thanks to reply: 1)SIP 2)yes. I've used the original 1.46 for SIP protocol Also your solution do not work. Are 2 days that I'm trying configurations and googling for this problem, but nothing! Always: LOG ON FAILED I've saw about problems with this phone, but my hope was that with the new firmware something could be solved. Thanks again. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: 1)What is the protocol you are using? SIP or IAX2? 2)Have you applied the correct firmware to the Phone? Pa168 phones are falwless when connecting to Asterisk. Start the configuration as asimple entry as under. I have added Port address and allowed codecs in the config below: [221] type=friend username=221 secret=secret context=local host=dynamic dtmfmode=rfc2833 nat=yes Port=5060 Disallow=all Allow=g729 Allow=ulaw Allow=gsm Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168S/AT320P Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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 -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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 -- .:FaberK:. ___ --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] Which asterisk-friendly cards are fax-capable?
I have no clear idea how many people actually use my software for fairly high volumes. There are now clearly many thousands successfully using it for modest levels of faxing. I have heard from a few people doing rather higher volumes than you. Other people have problem - I mean genuine problems, rather than the frame slips issues. I don't get enough feedback to really work out what it going on with those troublesome installations. What is the best way to determine if problems are genuine problems or are frame-slip issues? I have a dual xeon 3.0/2GB ram with a T100P connected to a PRI. We do not have a very high fax volume. Right now we recieve about 15 faxes per day, with each fax tending to be anywhere from 5 to 25 pages. (e.g. 75 to 375 pages/day) I have found 3 specific fax machines (all 3 are internal fax machines at our remote offices) that refuse to fax even a single page to spanDSP. 2 of the machines are HP machines, and the the 3rd was a brand I'd never heard of. I can attempt (and fail) to send from one of the 3 problem machines and then immediately send a perfect 25 page fax from one of our other machines. zttest shows 100% most of the time, but 99.987793%'s pop up in there sometimes. I'm guessing this is an indication of frame-slips. Do some fax machines just have better error correction than others? All 3 of the problem faxe machines belong to us, so if the problem does not sound like frame-slips, I can provide any kind of testing or logs that might help determine what the issue is. Eric ___ --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] Which asterisk-friendly cards are fax-capable?
Craig Guy wrote: I'm trying to figure out what an appropriate deployment model might be. Whether to have iaxmodem installed on the hylafax server with a switched ethernet connection for iax2 to the * server with the PRI, or to have the iaxmodem on the PRI * server and channel the tty comms across the network. You can try running the IAXmodem IAX channel over your switched network, but my recommendation would be to always run IAXmodem on the Asterisk system (to prevent even minute audio corruption). In my experience passing modulated fax audio over a small LAN has not been that big of a problem. Everyone that plugs fax machines into SIP ATAs (and even IAXys, I've heard) are a testimony to that. However, in those situations I think that either they have a very well-tuned network, a very low-traffic network, or the ECM capabilities and protocol error recovery features of their fax machines are managing to work around any audio corruption that may occur. I wouldn't recommend passing modulated fax audio over a UDP/IP network for businesses where those faxes are critical. As you may observe from the IAXmodem docs and patchset within the tarball, I have used IAXmodem in conjunction with termnetd+ttyd from the termpkg package. In my testing and small production usage with that configuration I have not had any severe problems with the tty or with any degree of data corruption occurring. However, I'm not yet convinced that the modem initialization, resetting, and other control handling that occurs on both ends of HylaFAX-faxmodem communications. In other words, I'm not yet certain that I've tuned my termpkg usage perfectly for use on high-traffic deployments where one call may arrive moments after the last one ended. If my concerns are confirmed and if there is no solution with termpkg to improve things, then I will have to create a busy-out AT command for IAXmodem that will tell the modem to return congestion until the busy-out setting is removed, and HylaFAX would busy the modem out during initialization and reinitialization cycles. I do this already with other DID modems where busying out a line is possible. I suspect that the latter might work ok over a WAN so I could have a central hylafax server with distributed * servers running iaxmodem at the far end of wan links (up to 100ms latency). I would suspect that you could run remote ttys over the internet and still use them for fax, yes... as long as IAXmodem is on or very close to the Asterisk server. The only issue is that I want to retain rxfax on the PRI * servers for incoming faxes. Lee, if I install iaxmodem on a * server for outbound faxing from hylafax, can I still use rxfax on the same server to receive faxes? If you're really so-possessed, yes. ;-) The only trick to watch out for is spandsp. Both rxfax and IAXmodem use spandsp, so you'd want to make sure that the version of spandsp that you're using is happy with both rxfax and IAXmodem. Lee. ___ --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
R: [Asterisk-Users] PA168S/AT320P
Why don't u attach the setup page of the phone ? Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di FaberK Inviato: giovedì 13 ottobre 2005 17.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] PA168S/AT320P Right now, but nothing changed. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: have you configured the STUN server on the phone to any one of the available stun servers like stun.xten.net? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 10:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PA168S/AT320P Hi, thanks to reply: 1)SIP 2)yes. I've used the original 1.46 for SIP protocol Also your solution do not work. Are 2 days that I'm trying configurations and googling for this problem, but nothing! Always: LOG ON FAILED I've saw about problems with this phone, but my hope was that with the new firmware something could be solved. Thanks again. 2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]: 1)What is the protocol you are using? SIP or IAX2? 2)Have you applied the correct firmware to the Phone? Pa168 phones are falwless when connecting to Asterisk. Start the configuration as asimple entry as under. I have added Port address and allowed codecs in the config below: [221] type=friend username=221 secret=secret context=local host=dynamic dtmfmode=rfc2833 nat=yes Port=5060 Disallow=all Allow=g729 Allow=ulaw Allow=gsm Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Thursday, October 13, 2005 9:35 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] PA168S/AT320P Hi all! I've got a problem with thia PA168S/AT320P telephone. I got 2 servers: one with SER and the other with Asterisk. All users are on SER and Asterisk is the gateway/voicemail. In these days I'm starting some tests using Asterisk accounts users. With this PA168S/AT320P, if I use it with a user from SER, it's ok but I can forget to use it with Asterisk users!!! I've also updated the firware at the 1.46 released the october 10th, but nothing changed. These are my user settings: [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local [EMAIL PROTECTED] callerid=221 221 accountcode=221 qualify=yes Any ideas? Thanks to all. -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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 -- .:FaberK:. ___ --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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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 -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or
[Asterisk-Users] PRI calls to Automated Attendants Dropped
I have 2 * boxes. 1 has 2 PRI's from the Telco, and a PRI to the 2nd * The other has ZAP channels to Channelbanks for endusers. If someone on the second box calls a Toll Free number (it probably doesn't matter that it is toll free) that is auto answered by an auto attendant (QVC, a Bank, the Airlines, Credit Card Companies) then the call gets dropped with in a couple of seconds of placing the call (the auto attendant barely gets started). Has anyone ever heard of this? I heard of people not hearing the auto attendant on some systems (not asterisk) because the channel isn't cleared or accepted (some sort of signaling related to these auto attendants). (maybe the same signaling that shuts off the audio on some PBX's is hanging up the *???). Any ideas/solutions? ___ --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] PRI stopped accepting calls
Hi, I have an asterisk box with a TE410P (quad pri) which has 3 spans in use, 1 and 3 to two different telcos, span 2 to a legacy Norstar MICS. Everything has been working fine for months, but early this morning, the 1st span stopped accepting incoming calls, but outgoing calls on this span still worked. Nothing in the logs indicates a change overnight, span reset, or anything else obvious, excep what I've noted below: - every hour, the b-channels were restarted without any indication of a problem - chan_sip stopped logging the 'retransmission' messages just before 6:13 b-channel restart - first failed incoming call at 7:40:50, 7:41:01 has log entry span 1 got hangup request - 8:13 b-channel restart logged extra Got restart ack on channel 0/20 span 1 with owner - 8:43 first successful outbound call on span1 -- no evidence or reports of any problems with outbound calls on span1 - 8:55 I started diagnosing problem - 9:03 successful inbound call on span 3 (different telco, different switch-type) Until 9:05 when I forced a restart, no inbound calls succeeded on span 1 -- all of them logged a 'hangup request' 11 seconds after the initial call setup. After searching the list archives, the only mistake in my config I've found is having both span 1 and 3 set as primary timing source (instead of having one of the set as secondary). /proc/zaptel/? shows that span3 is the timing source (since the restart, don't know about before). Could this be the source of the problem? Why wasn't I affected before? BTW, I've been restarting asterisk every 3 or 4 days as a preventative measure -- the current uptime for the process was less than 36 hours. Thanks for any light that can be shed -Gary ___ --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 channels marked with SIP_NEEDDESTROY but not being removed
I have been seeing the subject behavior on head for a few days now.. (been trying nightly builds to see if a bug causing this has been fixed) on a sip show channels I get a little of active channels that I can correlate calls to.. but I also have some dead channels listed that should no longer be there but still are anyway.. in the sip show channels list these channels are marked with a (d).. in looking at chan_sip.c the channel should be marked as SIP_NEEDDESTROY and should be removed in looking at the source.. The history for such a channel looks like: tranquility*CLI sip show history -363845771@ tranquility*CLI * SIP Call 1. Rx INVITE / 1 INVITE 2. CancelDestroy 3. TxResp SIP/2.0 / 1 INVITE 4. TxResp SIP/2.0 / 1 INVITE 5. TxRespRel SIP/2.0 / 1 INVITE 6. Rx ACK / 1 ACK 7. TxReqRelINVITE / 102 INVITE 8. Rx SIP/2.0 / 102 INVITE 9. CancelDestroy 10. Rx SIP/2.0 / 102 INVITE 11. CancelDestroy 12. Unhold SIP/2.0 13. TxReq ACK / 102 ACK 14. TxReqRelINVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INVITE 16. CancelDestroy 17. Rx SIP/2.0 / 103 INVITE 18. CancelDestroy 19. Unhold SIP/2.0 20. TxReq ACK / 103 ACK 21. TxReqRelINVITE / 104 INVITE 22. Rx SIP/2.0 / 104 INVITE 23. CancelDestroy 24. Rx SIP/2.0 / 104 INVITE 25. CancelDestroy 26. Unhold SIP/2.0 27. TxReq ACK / 104 ACK 28. TxReqRelINVITE / 105 INVITE 29. Rx SIP/2.0 / 105 INVITE 30. CancelDestroy 31. Rx SIP/2.0 / 105 INVITE 32. CancelDestroy 33. Unhold SIP/2.0 34. TxReq ACK / 105 ACK 35. TxReqRelINVITE / 106 INVITE 36. Rx SIP/2.0 / 106 INVITE 37. CancelDestroy 38. Rx BYE / 201 BYE 39. TxResp SIP/2.0 / 201 BYE To me it looks like the channel should indeed be removed as it is indeed dead.. but it remains in the sip show channels listing.. Is this a bug? Has this been run into before by others? Does anyone have a remedy for this? Is there perhaps a function that needs to audit periodically the sip channels list to expunge dead channels that should have been removed long ago but have not? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --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] PRI calls to Automated Attendants Dropped
Sounds similar to a problem I've seen with a slightly different setup Calls to certain AA/PBXs were not passing progress information beyond 10 seconds into the call. Can you check your logs for the exact amount of time after the setup that the call gets dropped? I'm guessing you'll see 10 or 11 seconds for most calls. The work-around is to make the asterisk box doing the relaying Answer() the calls before doing the outbound Dial(). On 10/13/05, Dave Wise [EMAIL PROTECTED] wrote: I have 2 * boxes.1 has 2 PRI's from the Telco, and a PRI to the 2nd *The other has ZAP channels to Channelbanks for endusers.If someone on the second box calls a Toll Free number (it probablydoesn't matter that it is toll free) that is auto answered by an auto attendant (QVC, a Bank, the Airlines, Credit Card Companies) thenthe call gets dropped with in a couple of seconds of placing the call(the auto attendant barely gets started).Has anyone ever heard of this?I heard of people not hearing the auto attendant on some systems(not asterisk) because the channel isn't cleared or accepted (some sortof signaling related to these auto attendants).(maybe the samesignaling that shuts off the audio on some PBX's is hanging up the *???).Any ideas/solutions?___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
