[asterisk-users] Spoofing CID
So who out there is aware of the FCC or FTC laws concerning spoofing caller ID for deceptive purposes? There's a collection agency out there who has my wife's name crossed with someone else's, and they are picking numbers from our area code to present themselves as when calling us (over and over and over.) I of course would like to turn this around on them as they refuse to believe who we say we are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spoofing CID
Yeah I'm thinking either homeland security or some other identity-critical legislation might be on my side here. On Thu, Jul 3, 2008 at 12:40 PM, randulo [EMAIL PROTECTED] wrote: On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear [EMAIL PROTECTED] wrote: So who out there is aware of the FCC or FTC laws concerning spoofing caller ID for deceptive purposes? There's a collection agency out there who has my wife's name crossed with someone else's, and they are picking numbers from our area code to present themselves as when calling us (over and over and over.) I of course would like to turn this around on them as they refuse to believe who we say we are. That sucks! Here's an older article about this seemingly common practice: http://www.securityfocus.com/news/9822 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chatterbug
Wow. How on EARTH do these people stay in business? Just running the law of averages and hoping it works out? $10 a month for unlimited routing through their 800 number seems like a risky gamble for them. On Nov 12, 2007, at 9:21 PM, Paul Hales wrote: http://www.oldskoolphreak.com/tfiles/voip/chatter_bug.pdf PaulH On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote: Does anyone know anything about the Chatterbug product? I can't tell if it's an ATA with a modem or some sort of LCR proxy or somesuch. Anyone? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chatterbug
Does anyone know anything about the Chatterbug product? I can't tell if it's an ATA with a modem or some sort of LCR proxy or somesuch. Anyone? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mixing Vars into Voicemail WAVs
Has anyone out there tried to mix the envelope metadata for voicemails into the audio payload that's stored by Asterisk? I would like to have the CID and Timestamp baked into the beginning of the WAV file, not just as text in the email itself. Thanks! -Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZOOM 5806 ATA
So in my ignorance I bought a Zoom 5806 ATA from Micro Center. It was cheap, what can I say? Anyhow, the docs are horrible, but the control panel is fairly straightforward. I can get it to register against Asterisk but I cannot get it to dial. Does anyone have a working configuration they can share? Thanks! -Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Marketing 101
Agreed. Highly-considered purchases like telco infrastructure are not as much a push as a pull sale. It's about being in the right place at the right time with all the right answers. Almost like buying a home. Since the turnover is SO long with core business process equipment, it's almost a beauty contest when the time comes around. A better analogy would probably be in luxury car buying. You need to look good, have a good feature set, be luxurious to drive, have all the right bells and whistles above and beyond basic requirements, and then of course have a track record of reliability and great service. Just my $.02 -- --- Robert Goodyear Managing Partner Brand Up LLC Knight West 949.542.7001 DIRECT 949.542.7010 FAX 888.272.6387 x501 [EMAIL PROTECTED] [EMAIL PROTECTED] --- On Apr 25, 2007, at 10:52 AM, SIP wrote: Businesses RARELY are in a position to choose new Telco systems providers. Oftentimes, that sort of decision is made by whomever leases them the office space, or was made once back in the beginning, and they've had no real reason to re-evaluate their service/provider. There are, however, plenty of Telco events where the providers hawk their wares and the installers tout their expertise. Cold Call/Networking/Word of Mouth are decent methods of getting your name out there as an alternative, but be prepared to run into a great many situations in which the system or provider they have 'works well enough' so they're not interested in changing. shadowym wrote: Thanks for the advice. Maybe I should clarify what I was asking. It's not so much the how but the what. What are people doing to get PBX Sales/Support business. I know how to get IT business but potential customers still see the Telco business as quite different and are used to using separate companies for that. What I was asking is how the traditional telco guys get new sales/support/consulting business. With IT it's usually a combination of cold call/networking/word of mouth. I'm hoping that Telco is the same but I never see any telco guys at networking events so I am thinking they cold call and advertise targeted at business owners. I'm not sure though. -Original Message- From: dave cantera [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 24, 2007 9:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Marketing 101 shadowym, best thing to do is talk to a lot of consultants, coaches, and marketing people... take the approach you do with learning open source only reverse it... instead of reading source (internal) ask people (external)... it is a big undertaking and the most important task you have... marketing is a bigger task than the technical (for a tech anyway) don't go it alone nothing happens without marketing (and sales)... marketing is *not* sales... daveC shadowym wrote: I have some general questions about marketing. Lot's of technical info but I was wondering how people are getting the business to begin with. I'm from the IT end of things but Telco is quite a bit different. Is cold calling still the way to go or networking? General stuff like that. Are there any resources on the web I can search for? Any suggestions would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snmp Monitor for asterisk boxes
Hello all, Witch snmp system do you use to collect info about their asterisk boxes, for example, uptime, downtime, max load, HD, free memory, asterisk status, ,etc? I use Nagios and the extension that logs in to the * manager interface. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail issue
Check for the existence of INBOX and OLD folders in the VM folders in /var/spool/asterisk/voicemailI messed around with something and found this out the hard way.Let me know if this works, as I am curious.-Rob.On Oct 10, 2006, at 12:58 PM, stan ford wrote:the last thing i was trying to do was change the default password to same as voicemail. i also tried reversing these changes but doesnt work. this is my log. i should probably mention that im running trixbox 1.21. when i connect to the voicemail system remotely, i enter the username, then a password and thats when this comes up. Core debug is at least 1 -- Executing Macro("Local/[EMAIL PROTECTED],2", "hangupcall") in new stack -- Executing ResetCDR("Local/[EMAIL PROTECTED],2", "w") in new stack -- Executing NoCDR("Local/[EMAIL PROTECTED],2", "") in new stack -- Executing Wait("Local/[EMAIL PROTECTED],2", "5") in new stack -- Executing Hangup("Local/[EMAIL PROTECTED],2", "") in new stack How low will we go? Check out Yahoo! Messenger’s low PC-to-Phone call rates.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] ftp server
Couple suggestions:On Win32, try Serv-U FTP. It's very reliable and supports a variety of protocols like SFTP and the like. Commercial software.http://www.rhinosoft.comOn Linux, I've fallen in love with FreeNAS, and not just because it utilizes the m0n0wall GUI. You can boot it off a USB stick, too, which I find super-cool.http://www.freenas.orgGood luck,-Rob.On Oct 8, 2006, at 11:29 AM, Gary Eck wrote: Yes, we don't have the proper skillset, either. Even though we are a reseller. We have standardized on Linux appliances called Snapgear . Of course, they are not integrated into Active directory, so there are lots of things that are harder to do. However, since they are not Microsoft or Cisco based, they are not the targets of many of the severe attacks on the Internet. We also have problems with installing all the security updates on the scores of Small Business Customers we have - they don't want to pay us for installing these updates - much less, paying us when a security update has side effects on their internal network. So, avoiding the whole Microsoft issue is a decent compromise in our situation. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 1:16 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp server Unfortunately I disagree with you about ISA, whilst it may cause problems through my lack of skillset from time to time the functionality it introduces and protects my network cant be beat at any price, being built into sbs 2003 is just a bonus. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gary EckSent: Sunday, 8 October 2006 1:37 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp server We also have SBS2003 - we initially had ISA running on it, but decided it was not worth the grief at our office. In the back of my mind, I was wondering if ISA was causing the problem. I had used Bulletproof in the past, since it has bandwidth throttling - and it just seems to work fine. Unfortunately, it does not run as a service - but I found another product that allows it run as a service. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 12:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp serverYep, using sbs 2003 here. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Gary EckSent: Sunday, 8 October 2006 1:17 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] ftp server We had some problems with the FTP server on our Windows 2003 server - we worked with it quite a bit, but just could not get the Pollycom's to access it. It did work with backup up our Switchvox PBX. We use Bulletproof FTP server in the Windows world. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dean CollinsSent: Sunday, October 08, 2006 12:46 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] ftp serverWhats the best ftp server to upload Polycom phone cfg’s from? I’m finding it a bit hit and miss using BTF server. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ftp server
Whoops. I totally generalized under the realm of Non Windows didn't I? Doh! On Oct 8, 2006, at 2:27 PM, Tzafrir Cohen wrote: On Sun, Oct 08, 2006 at 11:50:53AM -0700, Robert Goodyear wrote: On Linux, I've fallen in love with FreeNAS, and not just because it utilizes the m0n0wall GUI. You can boot it off a USB stick, too, which I find super-cool. http://www.freenas.org /me imagines some BSD fanatics readying their tridents and aiming at Robert :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HTTP Connection Closed on 7960 SIP
Very interesting. The port was open but there was an HTTP proxy entry in the SIP config still. Thanks! On Oct 6, 2006, at 9:21 PM, Aaron Daniel wrote: This happens if you have a logo_url configured for your phone and the phone can't access it. I'm guessing you don't allow 80 through the firewall to the server that's serving the image. -- Aaron Daniel On Fri, October 6, 2006 20:13, Robert Goodyear wrote: Anyone know why I get HTTP Connection Closed on the display of a 7960 running a SIP image? Only seems to happen when registering against my Asterisk box from the WAN. I have 1:1 NAT happening on my firewall. Phones function perfectly otherwise. TFTP working fine across the firewall as well. Odd! Thanks in advance. -Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HTTP Connection Closed on 7960 SIP
Anyone know why I get HTTP Connection Closed on the display of a 7960 running a SIP image? Only seems to happen when registering against my Asterisk box from the WAN. I have 1:1 NAT happening on my firewall. Phones function perfectly otherwise. TFTP working fine across the firewall as well. Odd! Thanks in advance. -Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation again ...
Aug 17, 2005, at 5:44 AM, Tom Hayden wrote: I have experienced pretty nasty echo on my PRI w/TE110P. The echo was only coming from other POTS lines, because cell phones already have echo cancellation, and other PBX's had the same. I resolved the problem by turning on the AGGRESSIVE option and it works fine now, and we haven't noticed a severe degradation in sound quality - most of my operators were just happy the echo was gone :) +1 here too: Uncommenting AGGRESSIVE_SUPPRESSOR and recompiling took care of 99% of my TE110P/PRI echo. -Rob. -- Robert Goodyear Brand Up LLC http://www.brand-up.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] Lock Extension
On Aug 18, 2005, at 3:07 AM, Stephen wrote: Hi All, How can I lock the extension in Asterisk? For example , my extension is 1000 and I am away for business trip. I want to lock my extension during my absence. Can it be done in Asterisk? regards, Stephen You could write a little script to mangle/unmangle your SIP context and then SIP RELOAD. You could assign it to a context called 'disabled' whose only valid extension matching therein is to that same macro to authenticate and change your context back. ___ 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] Suggestions for mainstream hardware compatible with TE411P.
On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote: I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or … any other brand and model that is known to work well with the TE411P ? Will an old Proliant do?I've built five PBXes on Dell Dimension 2600s that run flawlessly. They're P3 2.6GHz machines, so processor load stays super-low. Using a combination of TE110Ps and VoIP termination/origination, across ~35 users at each location on 7960s. Never missed a beat.I would consider a "consumer" box with a strong CPU over an old server, then spend your money on an ATA RAID card and mirror everything for disaster recovery.Hope that helps. ___ 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] Very complicated dialplans?
Using 'r' flags makes baby Jesus cry. Stop doing that. Excuse me? r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. Can you educate us all on the appropriate circumstances in which to use 'r'? Thx, -Rob. ___ 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] Some echo?
Robbie: I fought with echocancel and various parameters for a long time with little luck. Then I uncommented AGGRESSIVE_SUPPRESSOR and DISABLED the Fax/tone detection in in zconfig.h since we're not faxing via Asterisk. Recompiled and all echo disappeared. Hope that helps. -Rob -- Robert Goodyear Brand Up LLC http://www.brand-up.com On Aug 4, 2005, at 2:10 PM, Robbie Hughes wrote: I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. Most calls are fine, all incoming calls are fine, but I am getting echo on a significant number of outgoing calls. The person on the other side hears a perfect call, but the SIPphone side gets to hear themselves. It happens 100% of the time to some numbers (outgoing only), and only sporadically to others. Has anyone ever experienced this? the RTT to the phones from the server is less than 10ms and it is a 100mbit network with no traffic and cisco switches. zapata.conf attached below: Note: The commented out gain of +2 on outgoing seems to make no difference to the effect. Has anyone got any ideas? ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] group = 1,16 [channels] spanmap = 1,1,1 language=en context=from-pstn rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 ;txgain=2.0 txgain=0.0 rxgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn channel= 1-12 faxdetect=both ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone
On Aug 4, 2005, at 10:37 PM, Martin Kronstad wrote: Hi!Problem:I can’t hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound.My current setup is:Softphones/Hardphones(Location A) - Asterisk - Firewall/Nat - Internet - Firewall/Nat - Softphone/hardphone(Location B)A great guide is here:http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.htmlPay very close attention to the externip and localnet parameters that belong in the GENERAL section of SIP.conf-- Robert GoodyearBrand Up LLChttp://www.brand-up.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] So you all think VoIP sypply is warm and fuzzy
On Jul 18, 2005, at 3:13 PM, Michael D Schelin wrote: Here is a letter I sent them for my $150 paper weight. The forum is not a place to post ransom notes. You've added zero benefit to any reader here, nor to yourself, since you didn't actually ask a question in your email. ___ 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] Re: So you all think VoIP sypply is warm and fuzzy
On Jul 20, 2005, at 12:22 AM, Brian Capouch wrote: Michael D Schelin wrote: Real scary who You certainly have found an unusual way to promote your business. B. Kinda sounds like a schoolyard taunt, usually found near most lemonade stands, doesn't it? ___ 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] MOH Class in MeetMe
Is is possible to specify the MOH Class when defining a MeetMe extension? I tried exten = 300,1,MeetMe(300|M(class)) But that did not work. Thx, -Rob. ___ 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] MOH Class in MeetMe (Solved)
On Jul 14, 2005, at 11:17 AM, Robert Goodyear wrote: Is is possible to specify the MOH Class when defining a MeetMe extension? I tried exten = 300,1,MeetMe(300|M(class)) Replying to my own query, just in case anyone else is as dense as I am... exten = 300,1,SetMusicOnHold(confclass) exten = 300,2,MeetMe(300|M) I don't know why but for some reason I was convinced that setting the class here would not carry forward to the Conference's scope. Thanks for listening. -Rob. ___ 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] Skip Announcement Confirmation in MeetMe
Anyone know how to bypass the CONFIRMATION of the user announcement recording in MeetMe? While I like the please say your name to announce a user into a conference, I find it confusing and time consuming to make the user to press 1 to accept a recording they haven't even previewed. I'm not a coder, but I'd be happy to comment out the confirmation loop if someone pointed me in the right direction. Thanks. ___ 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 NAT + m0n0wall 1:1 mapping
I know a SIP client behind a NAT trying to peer with Asterisk behind another NAT is troublesome. Has anyone had any luck doing this by interfacing Asterisk to the WAN using 1:1 NAT translation to give it a public IP while still firewalled? In my instance I'm using m0n0wall, but this is a hardware-neutral question. Thanks. -- Robert Goodyear Brand Up LLC http://www.brand-up.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] Definitive CallerID Format and anonymous?
On Jul 8, 2005, at 12:43 AM, Jay Milk wrote: All, I'm currently only setting CID as a ten-digit number. Has anyone on this list tested caller-id delivery with various services? Is there *one* usable format (i.e. 1+10, or +1+10), or does it vary from provider to provider? Jay, FWIW the US standard for CLI is ten digits. I don't know if this has anything to do with your root question, but I thought I'd chime in here. I notice that one of my providers sends 1 plus ten for US calls, which is nonstandard and thus breaks CNAM lookups on the recipient's end via their PSTN provider. When one of my providers was sending the plus sign and eleven digits, it would break completely when that call was destined for an ATT cellphone. Hope that helps. -Rob. ___ 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] Teliax Passing Audio?
Is anyone having issues with audio being passed inbound via Teliax? Trying to isolate an issue here. Thx, -Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Providers Problems
On Jul 4, 2005, at 2:43 PM, Jimmy Smith wrote: you guys are so friggin funny.. We try. Meanwhile, you are SO illiterate; are you trying? ___ 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] #include not working with *1.0.9
On Jul 4, 2005, at 12:05 AM, [EMAIL PROTECTED] wrote: We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called extensions-phones.d and it contains a number of .conf files. In my extensions.conf file I have put a #include extensions-phones.d/*.conf in my [globals] context That happened to me in Jan or Feb of this year; just happened to be that on one particular day, the source I CVSed out had a broken * shell expander. I waited a day or two, redownloaded and recompiled and all was well. -- Robert Goodyear Brand Up LLC http://www.brand-up.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] #include not working with *1.0.9
On Jul 4, 2005, at 12:17 AM, Bryce Chidester wrote: Just a thought, but I seem to recall that in the dialplan, inlcude and other similar statements are not prefixed by the hash character (#). Try include = . -Bryce You're thinking of contextual includes, not filesystem includes -- which do use the hash. ___ 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] Sometimes yes - sometimes no (dialplan)
I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. I cannot make up a CLI entry ;-) There is nothing about it!!! As I said it is like it is not connected! Well if you say it's registered, then packets are getting to asterisk and asterisk is accepting them, and you've allowed that SIP client. So... if you say there's absolutely NOTHING happening when the phone dials, then it sure seems like the phone is bad -- again, assuming no event whatsoever is happening when you dial. What else have you done to debug this? Have you registered the phone directly against another * box? Have you registered another phone against this * box? ___ 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] play message to callee before connect toincomingcall
On Jul 2, 2005, at 1:00 PM, Roland Zagler wrote: sorry for the misunderstanding, robert! the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. Ah, now that's a clearer picture of what you're after. Perhaps you need to create a call file that then joins the two legs of the call afterwards? ___ 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] Sometimes yes - sometimes no (dialplan)
On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote: Robert Goodyear wrote: On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? ___ 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] Sometimes yes - sometimes no (dialplan)
I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? The CLI prompt does not show anything either. It is like the phone is not talking to asterisk at all. sip show users/peers does show the phone. ...shows the phone REGISTERED, yes? yes!!! ...yet no other information in the CLI or logs? C'mon, help us help you. The clue is in the question. ___ 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] play message to callee before connecttoincomingcall
the point is: during the caller is listening to the soundfile played to him the dialplan should continue to dial the sip phone 100 and after sip phone 100 is answered and the announcement file is played the caller should be connected to the sip phone 100. Ah, now that's a clearer picture of what you're after. Perhaps you need to create a call file that then joins the two legs of the call afterwards? yes, robert, but how do i join the two legs inside a call file or in the dialplan? i have experienced that call files can do a call to a channel and if this call is answered it can either be connected to an extension inside a context or call an application with parameters. roland Well, if you're already comfortable with AGI and realtime, I should think you could work something out with MeetMe, Conference or ParkandAnnounce. ___ 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] play message to callee before connect to incoming call
On Jul 2, 2005, at 10:23 AM, Roland Zagler wrote: Hello, i try to do the following: 1) call comes in on extension 999 2) caller should hear music (NOT MoH!!!) 3) a call should be initiated to SIP Phone 100 4) when SIP Phone 100 is answered, a sound file should be played to the user at SIP Phone 100 5) the incoming call (at extension 999) should be connected to SIP Phone 100 Rough pseudo-code follows, experiment and report your results to the list: 999 dial(SIP/100|20|m(soundfile)A(announcementfile)) OR 999 background(soundfile) 999 dial(SIP/100|20|A(announcementfile)) -- Robert Goodyear Brand Up LLC http://www.brand-up.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] play message to callee before connect to incomingcall
On Jul 2, 2005, at 12:55 PM, Mahmoud Badran wrote: try this one exten = 999,1,Answer() exten = 999,2,playback(~.mp3) exten = 999,3,dial (sip/100) exten = 999,4,playbackground(~.mp3) exten = 999,h,Hangup() not sure abt playbackground should be before the dial command or after Mahmoud: you don't pass file extension to the playback app, and there's no such app called playbackground. Plus the OP wanted the announcement to hit the callee solely. ___ 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] play message to callee before connect toincoming call
On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote: Thank you, Robert! The announcementfile plays well, but at Dial-option m i have to specify a MoH class, that is something i cannot use (as i wrote in my post). Noted, which is why I offered option two. Background command waits for a user input, but the caller should be connected to SIP Phone 100 after it has answered and the announcement has been played. But it doesn't REQUIRE input. Background completes when then sound file ends. Are you saying you want to move on to announcing the call to the callee as soon as it comes in while the caller is listening to the soundfile? I was following your sequential steps in your post, but if you intend to fork the process and be doing two things at once, then it's more complex. Before connecting to SIP Phone 100 the caller should hear a soundfile... wiki says nothing about an Dial-option to play a soundfile to the caller ;-( Sure it does... BACKGROUND. ___ 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] Sometimes yes - sometimes no (dialplan)
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote: I am confused about one of my installed server The dial plan seems to be ok, but sometimes NOTHING happens if I try to dial an extension (from X-Lite), next time it is done. X-Lite does not have a tone, nothing and does also have no time out. It seems it is not connected to the server. However, a sip show users / sip show peers shows that the phone is connected. SIP clients generate their own dialtone, so if you've got no tone, that sounds suspicious of a problem with the client itself. I assume you've debugged the problem by registering a hard SIP client on that server? -- Robert Goodyear Brand Up LLC http://www.brand-up.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] Computer to use
On Jul 1, 2005, at 4:03 AM, Eric Wieling aka ManxPower wrote: Robert Goodyear wrote: I'm sure you really only want to know about the absence of problems. From watching this list for 6 months it seems the SuperMicro products are most lauded and have exhibited no hardware conflicts. Various votes on Dell products, so you're probably best to stay away, even though I've got five installs with TE110Ps in them that have never missed a beat -- Dimension boxes, not PowerEdge. The SuperMicro Xeon board we tried failed miserably with both the T100P and TE110P. It had the ServerWorks IDE Chipset, which I suspect was the problem. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 Bummer! I thought I'd heard all good things about them... sorta like VoIP providers; as soon as everyone agrees things are OK, something goes awry! -Rob. ___ 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] Computer to use
On Jun 30, 2005, at 3:30 PM, Dovid B. Asterisk Users wrote: Hi, Already posted once but I need more feedback. What kind of servers is everyone using for asterisk and what problems have you ran in to ? Thanks. Dovid___ I'm sure you really only want to know about the absence of problems. >From watching this list for 6 months it seems the SuperMicro products are most lauded and have exhibited no hardware conflicts. Various votes on Dell products, so you're probably best to stay away, even though I've got five installs with TE110Ps in them that have never missed a beat -- Dimension boxes, not PowerEdge. -- Robert Goodyear Brand Up LLC http://www.brand-up.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 Asterisk documentation
On Jun 29, 2005, at 3:40 PM, harry gaillac wrote: Hello, If asterisk.org can't provide you documentations have a look here : http://www.digium.com/index.php? menu=product_detailcategory=softwareproduct=ABE I do hope some people understand my posts. Regards Harry Yeah, loud and clear. By the way, ever heard of a company called RedHat? ___ 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] Group/Broadcast Voicemail
On Jun 28, 2005, at 2:06 PM, Chris Stinson wrote: Were you guys able to figure this out? Robert Goodyear [EMAIL PROTECTED] wrote : On Jun 22, 2005, at 1:50 PM, Zen Kato wrote: Hi Robert, Let me guess... mailbox 5103 or 5203 were the last in the list to receive it? Every trials(1-6) I got only 51 mailboxes copied. My quick guess is 256/5(u0103 and xx03s)=51...1, so changing tmp[256] to tmp[4096] does not work. 'Pseudo-diagram' as you mentioned before(6/8/05) is desirable for expandability, but it also did not work. So what about the variable BASEMAXINLINE? Did you change that and recompile yet? Yes, I changed #define BASEMAXLINE on step 5(line 80) and step 6(line 82) and recomiled each case. I haven't had time to play with this. I posted over to DEV hoping someone who had their hands on that source had something to say, but nothing yet. -- Robert Goodyear Brand Up LLC http://www.brand-up.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] polycom soundpoint ip 300
On Jun 27, 2005, at 9:38 AM, Wilson Pickett wrote: Just so you know who you're dealing with: -- Forwarded message -- From: harry gaillac [EMAIL PROTECTED] Date: Jun 24, 2005 7:58 PM Subject: Re: [Asterisk-Users] polycom soundpoint ip 300 To: Wilson Pickett i piss on you Wilson Pickett Harry from France Wow, Wilson, what on earth did you do to attract said Gallic (Oh wow! I just put a joke within a joke there!) urinary secretions? ___ 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 say number but in french
On Jun 27, 2005, at 1:04 PM, David John Walsh wrote: Hello, does anyone know how to get the say number (say.c) agi application to work in french [assuming that I have the French voice files] Maybe Harry knows but hasn't documented it yet. Sorry, couldn't resist :-) ___ 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] callerid in forwarded call
On Jun 25, 2005, at 8:58 AM, E Fierro wrote: Hi, Do anybody knows how to display the original caller's callerid when transfering a call to another extension on that extension's phone? Usually the extension who is transfering the call would display as callerid. Thanks. SHOW APPLICATION dial Look for option o. -- Robert Goodyear Brand Up LLC http://www.brand-up.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] Exposing Zap Channels on Server A to be Used By Server B
On Jun 24, 2005, at 9:07 AM, Wiley Siler wrote: x-tad-biggerHello All,/x-tad-bigger x-tad-bigger /x-tad-bigger x-tad-biggerI remember there is a way to use two Asterisk servers and allow one to see a virtual trunk that makes it so server B can use the ZAP channels on server A./x-tad-bigger x-tad-biggerDoes anyone know where I can find this? I am racking my brain trying to remember the terminology./x-tad-bigger x-tad-biggerIt was like creating a 24 channel virtual T1 connection from server B to Server A that allowed server B to not have any ZAP hardware./x-tad-bigger x-tad-biggerAnyone know what I am talking about? I am searching the Wiki now but not hitting…/x-tad-bigger x-tad-bigger /x-tad-bigger x-tad-biggerSetup: /x-tad-bigger x-tad-biggerServer A has TMD lines and Voip.providers/x-tad-bigger x-tad-biggerServer B has only some extensions, needs to connect to Server A and use its ZAP channels /x-tad-bigger FWIW I've just been IAX2 trunking over to my other server with the TE110P in it; works very reliably and I can do a failover to VoIP if, say, all channels are busy or something else bad happens. It can also give me two-way (inbound AND outbound) failover with timeout forwarding from my VoIP provider, where if after xx seconds my external inbound IAX2 trunk does not pick up (either by design or ISP failure) the call is routed to my Cox DID, then internally IAX2 trunked across to my PBX. I'm sure all this can be done with the TDMoE method, but I was just throwing this at you so can make an informed decision. Robert Goodyear Brand Up LLC http://www.brand-up.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] Exposing Zap Channels on Server A to be Used ByServer B
On Jun 24, 2005, at 1:31 PM, Wiley Siler wrote: x-tad-biggerFrom:/x-tad-biggerx-tad-bigger [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] /x-tad-biggerx-tad-biggerOn Behalf Of /x-tad-biggerx-tad-biggerRobert Goodyear/x-tad-bigger x-tad-biggerSent:/x-tad-biggerx-tad-bigger Friday, June 24, 2005 9:51 AM/x-tad-bigger x-tad-biggerTo:/x-tad-biggerx-tad-bigger Asterisk Users Mailing List - Non-Commercial Discussion/x-tad-bigger x-tad-biggerSubject:/x-tad-biggerx-tad-bigger Re: [Asterisk-Users] Exposing Zap Channels on Server A to be Used ByServer B/x-tad-bigger On Jun 24, 2005, at 9:07 AM, Wiley Siler wrote: Hello All, I remember there is a way to use two Asterisk servers and allow one to see a virtual trunk that makes it so server B can use the ZAP channels on server A. Does anyone know where I can find this? I am racking my brain trying to remember the terminology. It was like creating a 24 channel virtual T1 connection from server B to Server A that allowed server B to not have any ZAP hardware. Anyone know what I am talking about? I am searching the Wiki now but not hitting… Setup: Server A has TMD lines and Voip.providers Server B has only some extensions, needs to connect to Server A and use its ZAP channels FWIW I've just been IAX2 trunking over to my other server with the TE110P in it; works very reliably and I can do a failover to VoIP if, say, all channels are busy or something else bad happens. It can also give me two-way (inbound AND outbound) failover with timeout forwarding from my VoIP provider, where if after xx seconds my external inbound IAX2 trunk does not pick up (either by design or ISP failure) the call is routed to my Cox DID, then internally IAX2 trunked across to my PBX. I'm sure all this can be done with the TDMoE method, but I was just throwing this at you so can make an informed decision. Robert Goodyear Brand Up LLC http://www.brand-up.com x-tad-biggerRobert,/x-tad-bigger x-tad-bigger /x-tad-bigger x-tad-biggerEssentually I want to be able to have Server B dial the extensions connected to server A as well as route calls to the outbound route on Server A./x-tad-bigger x-tad-biggerServer B will have little to no knowledge of what is on Server A. I just want it to dump the calls off./x-tad-bigger x-tad-bigger /x-tad-bigger x-tad-biggerFor some reason I keep thinking this was a PRI type of thing. Like there was a module that loaded up as a fake PRI that your Asterisk box could use to connect./x-tad-bigger x-tad-bigger /x-tad-bigger x-tad-biggerThanks,/x-tad-bigger x-tad-biggerWiley/x-tad-bigger x-tad-bigger /x-tad-bigger x-tad-bigger /x-tad-bigger Right, the TDMoE virtual zaptel configuration as pointed out by Colin Anderson's post. But I also saw the note about 2.6+ kernel issues, so I threw my idea at you. As far as dialing in to Server A's extensions, either plan them so they don't conflict, or prepend a digit to push calls into the right context on Server A when coming out of Server B. Robert Goodyear Brand Up LLC http://www.brand-up.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] Group/Broadcast Voicemail
On Jun 22, 2005, at 2:07 AM, Zen Kato wrote: Hi, I also changed as following sequences; app_voicemail.c 1. Line 3724 tmp[256] to tmp[4096] vm_exec 2. Line 3760 tmp[256] to tmp[4096] append_mailbox 3. Line 3796 tmp[256] to tmp[4096] vm_box_exists 4. Line 3290 tmp[256] to tmp[4096] vm_execmain 5. Line 80 tmp[256] to tmp[4096] #define BASEMAXLINE 6. Line 82 tmp[256] to tmp[4096] #define BASEMAXLINE I tried to copy to 99 mailboxes, but no luck, only could copy to 51 mailboxes. -- Executing VoiceMail(SIP/1021-6bd9, u010302030303040305030603 07030803090310031103120313031403150316031703180319032003 2103 22032303240325032603270328032903300331033203330334033503 3603 37033803390340034103420343034403450346034703480349035003 5103 52035303540355035603570358035903600361036203630364036503 6603 67036803690370037103720373037403750376037703780379038003 8103 82038303840385038603870388038903900391039203930394039503 9603 970398039903) in new stack (snip).. -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') Jun 22 17:15:20 NOTICE[11044]: app_voicemail.c:1244 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 22 17:15:25 WARNING[11044]: app.c:994 ast_lock_path: Failed to lock path '': File exists .(snip).. Jun 22 17:15:25 NOTICE[11044]: app_voicemail.c:1244 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Unable to create lock file: No such file or directory I would like to copy to 100-150 mailboxes for one CPU. I also need someone's help. Let me guess... mailbox 5103 or 5203 were the last in the list to receive it? Robert Goodyear Brand Up LLC http://www.brand-up.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] Group/Broadcast Voicemail
On Jun 22, 2005, at 1:50 PM, Zen Kato wrote: Hi Robert, Let me guess... mailbox 5103 or 5203 were the last in the list to receive it? Every trials(1-6) I got only 51 mailboxes copied. My quick guess is 256/5(u0103 and xx03s)=51...1, so changing tmp[256] to tmp[4096] does not work. 'Pseudo-diagram' as you mentioned before(6/8/05) is desirable for expandability, but it also did not work. So what about the variable BASEMAXINLINE? Did you change that and recompile yet? ___ 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] Extension Configuration Best Practice
On Jun 21, 2005, at 11:48 AM, Denis Galvão - iSolve wrote: On 21 de jun de 2005, at 14:18, Jay Milk wrote: |Rich is indeed correct, Asterisk does not yet support multiple |registrations for a single peer entry. Thus when you register |the previous registration is discarded and the new one is |used. Thus like he said, the last one that registered gets the call. And asterisk will never do that, because that's not how SIP works. Is there a way to just register the phone when user pickup the phone!? In this way we can have two phones regitered with the same context. Sure, if you want it to never ring. Seriously though, the whole purpose of the registration method is to authoritatively peer with a useragent. Just use alias extensions that then define a ring group of SIP agents beneath them. I outlined my method for this in: http://lists.digium.com/pipermail/asterisk-users/2005-May/107245.html -Rob. Robert Goodyear Brand Up LLC http://www.brand-up.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] Should I choose DSL @ 1.5 or a full T1?
On Jun 17, 2005, at 7:56 AM, Daryl G. Jurbala wrote: You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400 and I'll give you the amount I save over the next quarter. NPA-NXX is 215-862. Good luck. That sounds almost like Xeno's Paradox there... if you gave away the savings you still be paying the same amount thus half the savings would be...? Sorry, just had to inject some Friday afternoon humor onto the list. Seriously though, I was never able to get a T1 for that price anywhere myself until I moved to Orange County, CA. -Rob. -- Robert Goodyear | Managing Partner | Brand Up LLC 901 Calle Amanecer | Suite 150 | San Clemente, CA 92673 Tel: 949/468.0370 x501 | Fax: 949/468.0371 | Cell/SMS: 949/981.7301 http://brand-up.com | [EMAIL PROTECTED]___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nobody picked up in 30000 ms
On Jun 16, 2005, at 8:23 AM, Kumara Jayaweera wrote: Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1,IAX2/[EMAIL PROTECTED]/10094472239112|30|tr) in new stack -- Called [EMAIL PROTECTED]/10094472239112 What country code is that you're dialing? Robert Goodyear Brand Up LLC http://www.brand-up.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] Group/Broadcast Voicemail
On Jun 9, 2005, at 5:14 PM, Chris Stinson wrote:
Robert Goodyear wrote:
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change in app_voicemail.c in the function vm_exec
set the tmp[256] to be tmp[4096] in an earlier replay so I did.
static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just
looking at it and then where BASEMAXINLINE is called (linked list of
users) looks like it might pay off. Try messing with that constant
and see what blows up :-)
-Rob.
Well, since I don't know jack about programming I will try to cut it
down some :)
So... any luck? If you can't adjust that list of users in the dialplan,
let me know and I'll play with the code and recompile.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
On Jun 10, 2005, at 6:38 PM, Michael Welter wrote: Barton Fisher wrote: I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -- -- ___ 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 Where are you located? What CLEC gives you a T-1 for $290? FWIW I provisioned a PRI and a DS-1 for $300 each. Don't know if I'm getting a break for having a voice and a data circuit broken out from one fiber drop, but that's what I'm paying here in Orange County. Also, I had a business cable modem before, which was *allegedly* not shared for business customers (suspicious) and the throughput was a roller coaster, as was the latency. The DS-1 cleared all that up. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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] Group/Broadcast Voicemail
On Jun 9, 2005, at 7:33 AM, Chris Stinson wrote: Here's what it looks like Robert -- Executing VoiceMail(SIP/6153245827-0a2e, [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED] [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]8 [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]83 [EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]840 @mcdstores[EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED]845@ mcdstores[EMAIL PROTECTED]@mcdstores) in new stack -- Playing 'vm-intro' (language 'en') -- SIP/6153245805-d694 answered SIP/207.65.117.4-bf434468 Do you think there's any coincidence that exten 838, where you indicate the last vm is copied to, falls right around character 256 of that argument? I would experiment by temporarily shortening the contexts to q (for headquarters) and s (for stores) and trying again. That would shorten the argument you're sending to the vm app considerably and would give proof if this is or isn't the issue. Let me know... I'm very curious now! Robert Goodyear Brand Up LLC http://www.brand-up.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] More than one account from the same provider?
On Jun 9, 2005, at 6:47 AM, Chris Mason (Lists) wrote:
Try this... get rid of the friend and use the peer and
user defs.
Then pay close attention to which parameters apply to those two defs.
Change the [teliax] to something different, like [teliax-in]
and [teliax-out].
All that will do is separate the configs for incoming and outgoing, I
need
two completely separate incoming and outgoing accounts.
Chris: you've answered your own question then. You'd have to convince
Teliax to send a different authentication name to your server. That's
why I was trying to clarify whether you meant outbound or inbound.
Given that we're talking inbound, I feel you're stuck.
Teliax could theoretically allow users to have a specific auth name
(could be as simple as [TELIAX-{username}] ) that their switch DIALs
against, but we're delving into territory where six of us on the planet
would want this and couldn't even come close to ever making it cost
effective for them to make such a change to their code.
Right?
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] howto write CDRs on two mysql servers
On Jun 9, 2005, at 8:51 AM, Rosario Pingaro wrote: For redundancy I would like to write the CDRs on tow mysql servers. cdr_mysql.conf accept only one configuration [global], how to add a second host? Might be easier to add a second host as a replica server with the mySQL Administrator. Might lessen the load on Asterisk by not waiting on a second, remote connection. /rg Robert Goodyear Brand Up LLC http://www.brand-up.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] Group/Broadcast Voicemail
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change in app_voicemail.c in the function vm_exec set
the tmp[256] to be tmp[4096] in an earlier replay so I did.
static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments to
see if that is, in fact, the issue before mucking around with source
and recompiling.
/rg
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] Group/Broadcast Voicemail
On Jun 9, 2005, at 2:55 PM, Robert Goodyear wrote:
On Jun 9, 2005, at 12:45 PM, Chris Stinson wrote:
I was told to change in app_voicemail.c in the function vm_exec set
the tmp[256] to be tmp[4096] in an earlier replay so I did.
static int vm_exec(struct ast_channel *chan, void *data)
{
int res=0, silent=0, busy=0, unavail=0;
struct localuser *u;
char tmp[4096], *ext;
I guess it has to be changed somewhere else. It's on 4096 right now
under the vm_exec. Evidently it needs to be changed elsewhere.
Noted, but I was wondering if you could try to shorten the arguments
to see if that is, in fact, the issue before mucking around with
source and recompiling.
In the spirit of the aforementioned mucking around, it feels like
BASEMAXINLINE might be the culprit. I am NOT a C guy, but just looking
at it and then where BASEMAXINLINE is called (linked list of users)
looks like it might pay off. Try messing with that constant and see
what blows up :-)
-Rob.
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Re: [Asterisk-Users] Group/Broadcast Voicemail
On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote: So, anyone else have any ideas? I tried the below suggestion and it's still only sending out 20 of the 32 voicemails. C F wrote: did you recompile afterwards? by doing make clean make make install On 5/2/05, Chris Stinson [EMAIL PROTECTED] wrote: Still only doing 20 voicemails. Thanks for the suggestion. - Here's a weird idea. Can you put each group of 20 users into a distribution group whose distributOR is a member of a distribution group itself? Pseudo-diagram, assuming: 400 is the master VM broadcaster and 5600 through 5631 are your 32 users. exten = 400,1,VoiceMail(u401402403) exten = 401,1,VoiceMail(u560056015602...5619) exten = 402,1,VoiceMail(u562056215622...5639) Wonder if that would work? Robert Goodyear Brand Up LLC http://www.brand-up.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] More than one account from the same provider?
On Jun 8, 2005, at 6:15 PM, Chris Mason (Lists) wrote: I have had good success with my efforts to send faxes over voip using ulaw, surprisingly, and I want to move it from testing to reality. I have an account with Teliax, who have been very good. For voice I use g729 and ulaw, but for faxing I can only allow ulaw. However, Teliax only sets the codec preferences by account. I have another account, but I can't see a way to register two accounts with one server. Any ideas? Chris Mason Outbound or Inbound? If outbound (you said SENDing faxes above, so I'm guessing here) you're not registering, you're connecting via the HOST, USERNAME and SECRET in the context in IAX.conf, right? Robert Goodyear Brand Up LLC http://www.brand-up.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] Ringing a few phones
On Jun 8, 2005, at 7:19 PM, Shidan wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shidan Sent: Thursday, June 09, 2005 11:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Ringing a few phones I have a client requirement that multiple phones can be dialed, however they don't want the pstn phone to pick up automatically because of voicemail etc, nothing can be changed on the phones, how can I handle this requirement, by the way no zap channels are involved, all the pstn phones are behing another sip gateway. On 6/8/05, Jennifer Hales [EMAIL PROTECTED] wrote: If you want to dial a number of phones at the same time do exten = 5000,1,Dial(SIP/5000SIP/5001SIP?5002). The value is what does the job. Kind regards Jenn Hi Jen thanks for the info but I already knew that, what I want is for it to not get picked up by voicemail on one of the channels. dialing them in sequence is not an option either, and as I mentioned changing the settings on the actual phones isn't an option either. I remember there was an option for the user to hit * to accept the call but I think thats only with ZAP, anyone know of a solution to this problem or something similar for SIP. Shidan More details needed. If you cannot control the behavior of the phones behind the other SIP GW (as you described it) then your only option is to control the duration of ringing to just below the threshold of pickup on those phones. Also, what happens when one of those phones is busy? If it goes straight to VM then that'll blow the whole timeout trick. Robert Goodyear Brand Up LLC http://www.brand-up.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] Group/Broadcast Voicemail
Robert Goodyear wrote: On Jun 8, 2005, at 6:14 PM, Chris Stinson wrote: So, anyone else have any ideas? I tried the below suggestion and it's still only sending out 20 of the 32 voicemails. C F wrote: did you recompile afterwards? by doing make clean make make install On 5/2/05, Chris Stinson [EMAIL PROTECTED] wrote: Still only doing 20 voicemails. Thanks for the suggestion. - Here's a weird idea. Can you put each group of 20 users into a distribution group whose distributOR is a member of a distribution group itself? Pseudo-diagram, assuming: 400 is the master VM broadcaster and 5600 through 5631 are your 32 users. exten = 400,1,VoiceMail(u401402403) exten = 401,1,VoiceMail(u560056015602...5619) exten = 402,1,VoiceMail(u562056215622...5639) Wonder if that would work? Robert Goodyear Brand Up LLC http://www.brand-up.com Tried that. Didn't work. Chris: 1. How long is the line in your dialplan that calls the voicemail app? 2. Have you tried casting a variable containing the concatenated list of extens and then passing that to the vm app? -Rob. Robert Goodyear Brand Up LLC http://www.brand-up.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] Digium G729 licensing - is it worth the trouble?
On Jun 6, 2005, at 4:31 PM, Chris Mason (Lists) wrote: Digium did acquiesce and allow me to relicense the codec today, essentially they asked me how any times I would like to be able to re-license, I hope you answered, As many times as is necessary to ensure I'm continually able to contribute to the livelihood and viability of the Asterisk community by being a user/developer/contributor of Asterisk and a supporter of your paid add-on sub-product by _honestly_ licensing something I could circumvent with a two-second Google search. Seriously. Even big, scary Microsoft lets users pick up the phone and explain a reinstall circumstance to unlock a lockout. And they're not even dealing with an open-source community who are constantly tweaking, adapting, modifying, recompiling, rotating out hardware ad infinitum ad nauseum to help make Asterisk the best product in its class. /rg ___ 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] Don't know what to do if second ROSE component is of type 0x6
I've seen this in the list archives; nobody had an answer. Having dug through tons of OSI docs, I cannot figure out what a second ROSE component of type 0x6 even is, much less debug its origin or reason the libpri pri facility code hates it. Anyone? Ref.: PRI NI2 TE110P CVS HEAD as of 05/24/05 ___ 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] secretary function
Christian Hiller wrote: Hello, we got a SNOM 360 here and this gota TRANSFER button. With this i can transfer a call from my phone another one. But when i push this Button and transfer the call to another phone, i get kicked out. Now, every secretary first asks the chief if he is available or not - how can i implement this feature thx for any ideas ! On Jun 3, 2005, at 7:03 AM, Mike Holloway wrote: Christian, I don't have any specific answers about your particular SNOM device, but what you are wanting to accomplish is an attended transfer, instead of a blind-transfer. You should verify that the SNOM 360 is capabile of doing an attended transfer. Cisco 79xx series phones provide both blind and attended transfer modes. -mike Enable atxfer in FEATURES.CONF so it will work with any hardware. *8 is the default key but I remapped features more intuitively like: *7 for call pickup (Star P) and *8 for call transfer (Star T) -- makes it very easy for people to remember using the letter of the feature. /rg ___ 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] ring requested on channel 0/23 already in use on span
Anyone know what ring requested on channel 0/23 already in use on span means? Happened this morning and locked up all inbound and outbound calls. Had to shutdown -r to repair it. PRI TE110P CVS HEAD as of 05/24/05 Thanks in advance. /rg ___ 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] a simple call to my girlfriend
a simple call to my girlfriend Unfortunately, the technology does not currently exist to make that possible. -Sorry, couldn't resist. ;-) ___ 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] VoiPSupply Dot Com
On May 31, 2005, at 4:30 PM, Karl J. Vesterling wrote: Garrett, evidently there is some verbage to that effect on the site. But just to let you know, no other business that we've done business with requires anything like that. Not a one. Also worthy of note is that the purchase was not a credit card order, so I'm rather surprised your terms regarding credit cards would apply. In retrospect, I guess I should have spent the 16 hours browsing your site looking for the fine print instead of waiting for a prompt shipment. But, alas... We found someone that knows how to do business with businesses. Since you're compelled to send us evidence of your other business dealings, why don't you send the list some pictures of yourself with some completely unimportant politicians to further validate your sense of self-righteousness? Listen: you obviously have no understanding of merchant accounts nor business risk management in general, so there's no amount of explaining a seller's right to uphold any and all terms in efforts of mitigating said risk. Go back to your workbench and sniff some solder fumes. /rg ___ 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] Suppress Missed Calls 7960 SIP
On May 31, 2005, at 8:05 PM, Andy Hamilton wrote: On 5/31/05, Robert Goodyear [EMAIL PROTECTED] wrote: Does anyone know how to suppress the Missed Calls indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg Rob: Not sure how to (though I agree it would be handy). If anything, it would be a Cisco thing. Have you checked their website to see if the have any tips? -Andy Yeah, no such luck. I'm guessing it would require a Firmware hack, which is CERTAINLY out of my realm. Which begs the question: I wonder if/when anyone will attempt to write FW for IP phones in the same vein as the openWRT / Sveasoft crowd. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP Cisco - can't find P0S3-07-4-00.sbn
On Jun 1, 2005, at 9:38 AM, Ing CIP Alejandro Celi Mariátegui wrote: What I need to do? Rename/Copy P003-07-4-00.bin to P0S3-07-4-00.sbn Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] No, renaming won't work, as it's a signed binary. Plus S versus O designates the application type. The file came with your firmware download from Cisco; it should have included: OS79XX.TXT POS3-07-4-00.bin POS3-07-4-00.loads POS3-07-4-00.sb2 POO3-07-4-00.bin POO3-07-4-00.sbn ___ 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] Suppress Missed Calls 7960 SIP
Does anyone know how to suppress the Missed Calls indication -- perhaps on a per-line basis -- on the 7960 running SIP? Reason: I've configured a group of extensions to ring for inbound calls and it seems pointless to accrue missed calls on those line presentations. /rg ___ 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] VoiPSupply Dot Com
It's not the licenses, that's like 10% of the problem. One can always buy licenses... But the 16 man hours I wasted waiting for the across-town overnight shipment vastly outweighed the cost of the licenses. To mitigate risk, why didn't you ask to pick up the product in person? The main gripe is: It's the fact that the next-day delivery for across town was two days late. Add to that the fact that it was shipped to the Bill-To address and not the Ship-To address. Thereby causing me to bill out 16 hours of my time (which isn't cheap), for sitting on my hands. THAT IS THE PROBLEM! I don't have any feelings about Voipsupply.com one way or another, but you're really beating a dead horse here about something that should be common sense. Like Ronald Reagan said about the Soviets: Trust But Verify. Translation: get a tracking confirmation number before you travel and waste your valuable time next time. Sh*t happens with shipping no matter how diligent the supplier is. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How To Connect an IP phone with asterisk
On May 24, 2005, at 9:30 PM, SYED ADEEL ALI wrote: Assalam Alaikum I want to know how can i connect IP phone with asterisk... which config files, i need to configure... plz tell me stepwise ... i m new to asterisk n i just used softphones with asterisk http://www.asterisk.org/index.php?menu=support ___ 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] Affecting overhead with Runlevel?
This is a _very_ green question, but I am just beginning to explore and learn Linux. Have to admit I avoided it for years due to other obligations but discovering Asterisk has forced my hand. So: knowing that the X11 window GUI is a resource hog, is it appropriate to use the GUI to install and configure various components, then set RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this give the same effect as doing a minimal install or is the mere presence of the installed (yet not inited?) packages too heavy? Corollary: if Asterisk is running as ROOT, is there any benefit to booting at RUNLEVEL 1 to prune the overhead down even further? Or is that really only for debugging or administrative issues? Thanks, /rg ___ 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] Affecting overhead with Runlevel?
On May 21, 2005, at 10:29 AM, Johnathan Corgan wrote: Robert Goodyear wrote: So: knowing that the X11 window GUI is a resource hog, is it appropriate to use the GUI to install and configure various components, then set RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this give the same effect as doing a minimal install or is the mere presence of the installed (yet not inited?) packages too heavy? That would work fine. You could still log in to the console and run the Asterisk console (asterisk -R) to watch things work, which is instructive. Noted. To clarify, will dropping back to runlevel 3 still ensure a smaller set of processes that would be as non-intrusive as if I had installed Linux with console/command line support only or would there still be stuff hanging around that's inextricably there because I had _at one time_ installed and run the GUI? thx /rg ___ 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] Affecting overhead with Runlevel?
On May 21, 2005, at 3:46 PM, Johnathan Corgan wrote: Robert Goodyear wrote: Noted. To clarify, will dropping back to runlevel 3 still ensure a smaller set of processes that would be as non-intrusive as if I had installed Linux with console/command line support only or would there still be stuff hanging around that's inextricably there because I had _at one time_ installed and run the GUI? No, runlevel 3 typically doesn't include any graphical console processes. Having them installed but not running only wastes disk space; there would be no difference to Asterisk from a CLI only installation. Outstanding. Thanks for your time. So that's the answer to my question; ensuring that unneeded stuff will be dormant after I'm done. /rg ___ 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] Anyone done the Cisco 7960 FW migration path programmatically?
Has anyone out there scripted the rollthrough migration of the Cisco firmware? It would be fantastic if there was an app that would generate a set of templated .CNF and XML files based on the MAC addys entered, then control and present your .BIN images through TFTP. It could then also send the reboot signal too, walking through the oh-so-ridiculous path from 3.2 (which every 7960 I've bought to date seems to ship with) through whichever image you've bought. If my scripting/programming skills weren't so weak I'd try myself. /rg ___ 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] CallerID name lookup AGI script
On May 20, 2005, at 3:48 AM, Chris Coulthurst wrote:
Well I installed this script on to my system (a few hiccups with php 5
but its not erroring anymore).
Still not getting any callerid info to pass to my polycom 500 screen.
Could it have anything to do with the fact that the number is prepended
with a +1 on the screen? Teliax sends the +1NXXNXX on the
number.
Is that being stripped by the agi script when it queries 411 and
google?
Or am I just a dumb fart no quite getting what I'm doing? (this is the
likely case!)
P.S. If anyone has a suggested script to remove the +1 from the
number,
it would be helpful in other areas as well...
I always conform the number before passing it anywhere. From Teliax I
do this:
exten = s,1,Answer()
exten = s,2,SetCallerID(${CALLERIDNUM:2})
exten = s,3,AGI(callerid.agi|${CALLERIDNUM})
Hope that helps.
/rg
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Re: [Asterisk-Users] Stange question...
On May 20, 2005, at 8:08 AM, Mark Johnson wrote: Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark What kind of clothing was she wearing? (Static electricity and plastic phones, you know?) Hope she's being nice about it, btw. I've had employees (try to) sue me for less ;-) /rg ___ 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] Boosting Shared Internet Bandwidth for Asterisk
On May 20, 2005, at 8:11 AM, chawki hammoud wrote: --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: You want Quality of Service. Google around, and then look at http://www.mixdown.ca/~andrew/dump/rc.tc. It's what I use and it seems to work very well. Could you please tell me where and how to install it Thanks. GOOGLE. LEARN. DEPLOY. You need a primer in IP networking before you endeavor to play with packet shaping or you'll be stabbing in the dark. You also need to ascertain whether or not it will be a complete waste of time if/when your provider completely ingores QoS. ___ 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] Road Warrior phone config
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote: The issue would be voicemail. He would want only one voicemail account to be accessed via the messages button. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 Infinite SIP account(s) could roll over into one VM account. I append digits to the end of an alias extension to define the various devices a call will ring to. Then they roll back up the that exten's VM acct. /rg ___ 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] Road Warrior phone config
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote: The issue would be voicemail. He would want only one voicemail account to be accessed via the messages button. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 Infinite SIP account(s) could roll over into one VM account. I append digits to the end of an alias extension to define the various devices a call will ring to. Then they roll back up the that exten's VM acct. /rg On May 15, 2005, at 1:19 PM, Chris Mason (Lists) wrote: Can you give me an example of the conguration used to do this? Pseudocode follows: exten = 456 ; Chris Mason exten = 4561 ; CMason Office exten = 4562 ; CMason Cell exten = 4563 ; CMason Home exten = 4564 ; CMason remote office 2 exten = 4565 ; CMason remote office 3 exten 456 1 dial 45614562456345644565 r 20 exten 456 2 voicemail u456 NB: assumes devices OUTSIDE your control (e.g. Cell, etc) don't (a)nswer within 20 secs and thus break the call flow. But you were talking about all SIP extensions logging into the same asterisk server, so you should be OK so long as none of the 456x entities do anything but ring. Make sense? /rg ___ 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] Road Warrior phone config
On May 15, 2005, at 12:27 PM, Chris Mason (Lists) wrote: The issue would be voicemail. He would want only one voicemail account to be accessed via the messages button. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 Infinite SIP account(s) could roll over into one VM account. I append digits to the end of an alias extension to define the various devices a call will ring to. Then they roll back up the that exten's VM acct. /rg On May 15, 2005, at 1:19 PM, Chris Mason (Lists) wrote: Can you give me an example of the conguration used to do this? Pseudocode follows: exten = 456 ; Chris Mason exten = 4561 ; CMason Office exten = 4562 ; CMason Cell exten = 4563 ; CMason Home exten = 4564 ; CMason remote office 2 exten = 4565 ; CMason remote office 3 exten 456 1 dial 45614562456345644565 r 20 exten 456 2 voicemail u456 NB: assumes devices OUTSIDE your control (e.g. Cell, etc) don't (a)nswer within 20 secs and thus break the call flow. But you were talking about all SIP extensions logging into the same asterisk server, so you should be OK so long as none of the 456x entities do anything but ring. Make sense? /rg On May 15, 2005, at 1:59 PM, Chris Mason (Lists) wrote: I know that part, dialing more than one extension and sending all voicemail to the same extension, but recovering voicemail is the hard part. On the Sipura and, I believe, on the Polycom, you configure the voicemail extensio, e.g., 8500, and the mailbox is derived from the extension yhou are dialing from. How do you get around that? Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 You serious? I typed all that and you were asking about RETRIEVING vm all along? Wow, I must be really dense today. So: don't pass calleridnum to extension 8500. Or configure a different voicemail retrieval exten for roaming users and pass null to voicemailmain. Or, even better, scrape off and discard the fourth extension digit when parsing calleridnum and handing to voicemailmain. /rg ___ 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] Callerid on PC and more
On May 15, 2005, at 5:29 PM, Anton Krall wrote: Guys. This is a good one... Is anybody doing callerid on the PC? What are you using besides yac or things like that? And are you using some CRM like Goldmine with it? Good huh? Yes, I touch userrecords in my SugarCRM implementation when an in- or outbound match is found. Assuming a match, I pop up the record on a user's PC to encourage a conversation log or disposition report against that record. Also, the evidence of a call event against an existing record with a lack of an associated convo log tells us our employees aren't faithfully tracking customer interaction. /rg ___ 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] Callerid on PC and more
|-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Robert Goodyear |Sent: Domingo, 15 de Mayo de 2005 08:42 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Callerid on PC and more | | |On May 15, 2005, at 5:29 PM, Anton Krall wrote: | | Guys. | | This is a good one... Is anybody doing callerid on the PC? What are | you using besides yac or things like that? And are you using |some CRM | like Goldmine with it? | | Good huh? | | | |Yes, I touch userrecords in my SugarCRM implementation when an |in- or outbound match is found. Assuming a match, I pop up the |record on a user's PC to encourage a conversation log or |disposition report against that record. | |Also, the evidence of a call event against an existing record |with a lack of an associated convo log tells us our employees |aren't faithfully tracking customer interaction. | |/rg | On May 15, 2005, at 6:52 PM, Anton Krall wrote: How are you popping up the screen on the users pc side? A listener on each client that calls an internet UA on significant events. Have you tried goldmine? No. I prefer more control of the deployment. Which is probably my own version of consultant-speak for I prefer to take more profit. ___ 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] POE hub
On May 15, 2005, at 7:19 PM, Chris Mason wrote: I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason You could wire your own. It's simply pins 4,5,7 and 8. If you do, beware the attenuation of a long ethernet run. That said, spending the time to wire up 16 ports may make a refurbed Cisco sound cheaper. /rg ___ 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] Callerid on PC and more
On May 15, 2005, at 10:44 PM, Waldo Rubinstein wrote: On May 16, 2005, at 1:32 AM, Robert Goodyear wrote: A listener on each client that calls an internet UA on significant events. I suppose by this you mean some sort of client software installed on the client PC that listens to events targeted at a particular port this software is listening to. If this is the case, how do you make Asterisk communicate with this client software? Listening for new rows in the mySQL db that Asterisk is writing events to. /rg ___ 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] Is there a product to simulate a PRI trunk?
Does anyone know of a way to simulate the signaling of a PRI trunk for testing/setup purposes? I realize this may be a rather naive question, but I was wondering if you could take a TE110, for example, and using a crossover cable (or not?) and some means of emulating the NI2 signaling protocol connect it to another TE110 on another machine to test and verify an installation before the telco comes out to provision our new trunks. Regards, /rg ___ 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] Is there a product to simulate a PRI trunk?
On May 13, 2005, at 12:39 PM, Chris A. Icide wrote: You should note that while this works technically, it won't catch any issues that you may experience when connecting to other PRI switches. In other words, two asterisk servers connected back to back with a T1 cross-over cable won't tell you that Asterisk's NFAS code doesn't work with Lucent 5ESS switches, or that Sangoma's code pre-firmware v1.1 and pre-driver beta6 versions won't bring up the D channels when connected with certain switches. Everything will work between two asterisk boxes perfectly. Right. Which is why I wanted to figure out if there was a way to emulate the particular signaling protocol; in my instance, NI2, to ensure I'm doing all the right things on my end based on what I tell the simulated telco end to impersonate. ___ 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] Is there a product to simulate a PRI trunk?
There is no way, with an asterisk box to emulate a particular vendor's switch. You can always change your signalling type and present different signalling protocols (asterisk supports quite a few), but you will never be able to emulate a switch from a different manufacturer, and then of course you have different firmware and model versions to deal with as well. I can say from experience that connecting two asterisk boxes back to back will get you 80% of the way there, when testing for functionality, but you won't know if you are going to run into a problem until you connect it to the actual switch hardware you plan to connect. Noted. OK, thanks to everyone for the responses. I'll set up for NI2 and hope for the best when the LEC brings the trunk up. Regards, /rg ___ 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] Cisco 7960 Can't be unlocked
On May 12, 2005, at 9:51 AM, Timothy R. McKee wrote: Those are SCCP based phones. move the cursor to option 3, but do not press select. press **#, then press select. You should see the padlock icon with an unlocked appearance. press 32 and see if you have a YES option (alternate TFTP). If so press yes, then go to option 8 and edit the ip address. The phone sometimes locks itself in the middle and I have to start over. tim FWIW: I've had some where the default password is 'cisco' /rg ___ 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] What do you name yours
I use tree names, alphabetically, e.g.: Ash, Birch, Cedar, Dogwood, Elm, Fir et al. Never had anyone asking me how to pronounce any Sci-Fi arcana that way. No offense meant to Sci Fi zealots, of course. ___ 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] Review Outgoing VM Messages
On May 6, 2005, at 11:02 AM, Christopher Jacob wrote: Hey All, I had a user ask how to go in and listen to her current outgoing messages. I must confess, I can't figure out how to. Any ideas? ~c Call herself? ___ 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] CNAM lookup: new method for Caller ID Name delivery
Other notes: The clever integrator of this application will save themselves some lookup $ by caching the responses from the database into their own database, along with a datestamp. Perhaps if an entry is 90 days old, the system will re-lookup the entry in the Accudata database but otherwise will present the memorized answer. (Hint: the caller ID's of your inbound call pool is probably 80% redundant) I wonder if CallerID names are in the public domain or considered public record? ___ 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] Audio cut off at beginning of call
On May 3, 2005, at 6:32 PM, snacktime wrote: On 5/2/05, Robert Goodyear [EMAIL PROTECTED] wrote: On May 1, 2005, at 11:39 AM, Gene Naden wrote: When we call out from our Asterisk system we consistenly lose the first roughly 1500 milliseconds of the audio from the destination. This is easiest to demonstrate with a recorded announcement. In other words, Hello for example is missing. We are calling over the PSTN via a voice T1 line. We are using the stable cvs from about April 1. I searched lists.digium.com but did not find anyone with this problem using the PSTN. Does anyone have any ideas? Same here, via VoIP. I reported it to the list a while back: http://lists.digium.com/pipermail/asterisk-users/2005-February/ 088514.html If you're getting it via ZAP and I'm getting it via VoIP, sorta starting to sound like a setup issue on the Asterisk side, doesn't it? I have had this same issue also on SIP and IAX calls, but it varies provider to provider. Last time I checked I had this issue with livevoip and teliax, but not with voicepulse. Which is curious because you had this with voicepulse right? Maybe they fixed this problem and the others just haven't caught on yet? It might be time for me to do another QA session. It's been a while since I did some A/B testing across my providers. FWIW I use Teliax, VP, VoipJet, SimpleTelecom and I have a few minutes to burn off of sixtel if they're still in business. I'll let you know what I discover. /rg ___ 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] Digium MOH
On May 3, 2005, at 6:46 PM, Matt Riddell wrote: Chris Mason wrote: Why not? Because you have not licensed the file for broadcasting across your telephone network. How many other people are there here that write music? Would there be any interest in creating a pool of music for Asterisk? Would there be any chance of creating a GPL exception for them if we donated them? I have rather a few songs, mostly in the trance/psytrance genre but also dub and DnB. Ideas? Good idea. Might be an interesting niche to fill by creating a selection of dropout-tolerant (and smoothly-compressible) seamless loops. IIRC I had some nice results using SoundEdit32 or perhaps it was BIAS Peak's loop factory on Mac OS X. Did a nice job of pattern matching to seam a selection cleanly. /rg ___ 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] Audio cut off at beginning of call
On May 1, 2005, at 11:39 AM, Gene Naden wrote: When we call out from our Asterisk system we consistenly lose the first roughly 1500 milliseconds of the audio from the destination. This is easiest to demonstrate with a recorded announcement. In other words, Hello for example is missing. We are calling over the PSTN via a voice T1 line. We are using the stable cvs from about April 1. I searched lists.digium.com but did not find anyone with this problem using the PSTN. Does anyone have any ideas? Same here, via VoIP. I reported it to the list a while back: http://lists.digium.com/pipermail/asterisk-users/2005-February/ 088514.html If you're getting it via ZAP and I'm getting it via VoIP, sorta starting to sound like a setup issue on the Asterisk side, doesn't it? ___ 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: SV: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
On Apr 25, 2005, at 10:55 PM, Ronald Wiplinger wrote: Hi Ronald, What happens in your Asterisk box when you press the Speed Dial number in IPS? Can we make it so that you FIRST answer below questions, please? | | Let's try it together: Ronald: wow. Take a breath before you torch a generous developer. IPS works like a charm for me in every way. Seriously, /rg ___ 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
