RE: [asterisk-users] Hotels...
PMS is the correct term for the hotel billing systems. Property Management System. Problem is that they are all proprietary interfaces and it is very hard to get the major companies to work with you. I've done so in the past. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Monday, August 07, 2006 8:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hotels... Interesting you said PMS? here is the definition: http://en.wikipedia.org/wiki/PMS On 8/7/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Ideally you'd get billing to work by integrating directly with the property management software. Most of the big PMS systems, such as SMS, LMS, and FRS, have custom serial drivers written for them that interface with the PBX and related systems. The PMS software is responsible for activating long distance on the phones, adding/removing voicemail boxes, and collecting billing records. It may also do really complicated things like suiting. I don't think you're going to be able to get any of that. For that reason alone, are you sure Asterisk is the right solution? Maybe a little Mitel system would work with their software? Okay, now assuming you have a nice GUI to rebuild mailboxes, you'll have to decide if it's worth restricting/unrestricting phones for long distance. Keep in mind housekeeping staff may like to make international phone calls. It may be easier to just sell calling cards and open up the lines for free local calls (usually local calls incur a surcharge of $1 - $1.50.) Are there conference facilities involved? Do you need special pricing for provisioning lines there? I like the idea of using those Audiocodes boxes, but will fax services work with them? In theory I think they do, but we've had problems passing data over them. Can the Audiocodes boxes drive message waiting lamps? I can't remember, but you'll need that. Wake up calls? Asterisk supports it (Trixbox has a nice implementation), so be sure to test that. Is this a multiproperty hotel and will you need to support 911 to different buildings? --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Monday, August 07, 2006 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hotels... I have to bid on a hotel contract, but there are some things I don't know how to do -- but clearly Asterisk has been used by hotels before, so I figure someone on here must have some answers: 1) While the majority of the phones will be SIP, there will be a couple hundred analogs (due to wiring logistics); what should I terminate them into? 2) Phone activation at check-in/phone de-activation and billing at check-out. Are there GUI tools for this, or should I write my own back/front end? 3) Anything else that those familiar with hotels have bumped into that might not be obvious at the outset? Thanks! -Ken Ken, Long time no see on the list welcome back. 1) The best thing would be is to get a channel bank. Xorcom has one that I believe works over USB though never tried it so I cant comment on it. 2)I dont think there is any software out there for hotels per say but there has been talk about working some of the open source billing programs out there in to a custom app. The only reason why I would go for writing your own is A)You have more control. You can build it for your own custom needs for the ground up. B)People have asked about it before. While I dont know the market size I am sure that you can resell it once you are done. Dovid ___ --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 __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] will a firewall slow down asterisk?
Any network device (ie: switch, router, firewall) will add a small amount of latency. To test the latency your firewall adds, you could simply try to do a ping www.google.com, directly in front and behind the firewall, and look at the ms response times. Cheers, S. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Langley Sent: Wednesday, August 10, 2005 2:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] will a firewall slow down asterisk? Hi there I am in the process of setting up a production Asterisk server, which will mainly be used for meetme conferencing. I am considering running a firewall, but wondering whether this will slow Asterisk down if all packets are being scanned. Any ideas? Many thanks Steven Langley ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Sometimes goes into a restart loop.
I've noticed with the CVS build I have, if I restart * it shuts down and then restarts and then shuts down and then restarts until I reboot my system. Anyone else have this problem? S. smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] test message - ignore me
Me neither.. but just started receiving now. WEIRD. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Monday, August 01, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] test message - ignore me Haven't seen email since the 29th.. just testing. smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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 is the problem with gmail and the list.
I have no spam lists. :P It died for many people I know. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit Sent: Monday, August 01, 2005 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] what is the problem with gmail and the list. I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and click Not spam hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] IPP transcoder compiling question
Check out http://aussievoip.com.au/wiki-G723-1-Install How to G723 for * -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Apu Islam Sent: Wednesday, July 27, 2005 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IPP transcoder compiling question Just downloaded and installed Intel's IPP transcoder for G.7231 and G.729 . However, once I am done with by Build.sh, I get the output g723encoder and g723decoder binariers on my bin folder. No codec_g723.so single file. What am I missing ? What should I change on Makefile ? regards, -apu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Get older CVS version
Try a -D 2005-05-29 16:47 S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Baranowski Sent: Thursday, July 28, 2005 12:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Get older CVS version Well the first time I sent this email was two days ago, yes I know that the internet headers may say that they where sent in short period of time today but they where not. If you send an email and it does not show up to the list and it has not come back or you don't get a delayed delivery what would you do? I have been having this issue ever since I have joined this list and I don't seem to have this issue with any other list on am on. Matter of fact one of my other postings has not shown up yet either and it has been 2 days and one back 4 days. As I am writing this I am wondering if this will even make it. Since I have never worked with CVS before I was just looking for some guidance not to get slammed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Wednesday, July 27, 2005 11:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Get older CVS version On Wed, 2005-07-27 at 16:20 -0700, Rick Baranowski wrote: Is there a way to get a certain build date of asterisk from the CVS? I need to get this build date ?Asterisk CVS-v1-0-04/29/05-16:47:42? If so what would be the command? Sending the same question twice within 1 hour will not help especially when during the intervening time you could have typed :- $man cvs which is what I'm going to tell you now. With man and google you can actually _learn_. -- Dave Cotton [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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Klicking sounds in background
IDK if this might help :-) http://www.voiptroubleshooter.com/diagnosis/usersymptoms.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jochen Witte Sent: Thursday, July 28, 2005 2:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: AW: [Asterisk-Users] Klicking sounds in background No - the clocking is OK. It works correctly with different SIP stacks. When I use voipong to grab the sound, I can hear the klicking in all cases: when dialing in via PSTN or by IAX / SIP. So the klicking is not on the PSTN side, but on the RTP side. The mysterious thing is, that You can only hera it, when dialing in via PSTN. Has anybody any idea? Jochen -- Jochen Witte email: [EMAIL PROTECTED] web: http://alpha-lab.net -Ursprüngliche Nachricht- Von: Peter Childs [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 28. Juli 2005 03:42 An: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] Klicking sounds in background Your ISDN clocking is slipping (or not sync'd). Your digium PRI needs to clock off the ISDN. See zaptel.conf on the Wiki and set something like... Span=1,1,... (the second '1' is important.. Ie 'use as primary sync source') http://www.voip-info.org/tiki-index.php?page=Zaptel.conf+span+sintax -- Peter Childs NEC Business Solutions Ltd Ph:61-8-8301-4908 Mb:61-4-0819-7693 IM: pjcinaus (yahoo) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jochen Witte Sent: Wednesday, 27 July 2005 10:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Klicking sounds in background Hello, I have an IVR with Intel HMP SIP stack, which is a peer behind my Asterisk box (Asterisk 1.0.7, Digium PRI). When dialing in via PSTN, there are klicking sounds in the background, which do not appear, when dialing in via SIP (using Asterisk as pbx). The issue does not seem to be an alaw/µlaw problem. I tried trunking two Asterisk boxes via IAX and then call via two asterisks, but the same effect appears. Whenever there is PSTN involved, I have these klicking sounds, when there is no PSTN, everything works correctly. The setup works great with different SIP peers (others than the Intel...) Anyone has an idea? Best regards Jochen -- Jochen Witte email: [EMAIL PROTECTED] web: http://alpha-lab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.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 smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Public phone
Check out: http://www.voip-info.org/wiki-VOIP+Payphones -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: Thursday, July 28, 2005 4:03 AM To: Asterisk-Users Subject: [Asterisk-Users] Public phone A client wants to put phones in a semi-public place, using a Calling Card solution. What kind of hardware is suitable? I'm looking for a wall mounted booth, a SIP phone that can't easily be broken, but I might just use cheap analog phones and a channel bank. What do you suggest for the calling card software? This installation will be outside the US. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [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 smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] QoS windows client - cFosSpeed
I know this isn't directly related to * but I found it works very well in my voip environment. Check out cFosSpeed @ www.cfos.de. It gives you QoS based on applications and also seems to have increased my network throughput. Cheers, S. smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] super high bandwidth codec
I dont know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the quality sound better. Maybe its your SIP client/hardware phone that is giving you troubles. Storm. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Sunday, July 24, 2005 8:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] super high bandwidth codec Ive just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Phone manual..
You might want to try this group out: http://groups.yahoo.com/group/pa1688/ Most of these Chinese phones are using the pa1688. Cheers, Storm. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Wong Sent: Friday, July 15, 2005 12:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phone manual.. There is no brand on the phone.. it is from china. [EMAIL PROTECTED] wrote: On 7/15/2005, Bill Wong [EMAIL PROTECTED] wrote: Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? Might help to tell us the VoIP Phone you are using. Not all come capable of doing these things without some work. Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Video phone settings???
I found the problem was with eyeBeam when I had more than one video codec enabled. Try on eyebeam to only have h263p enabled. Does the video appear in the Echo test? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald_Wiplinger Sent: Monday, July 11, 2005 12:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Video phone settings??? I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic canreinvite=yes context=from-sip callerid=Ronald Wiplinger 6003 dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=h261 allow=h263 allow=h263p Tests on 7/11/2005 Eybeam to 8770 both screens are black!!! e*CLI -- Executing Dial(SIP/6003-94ec, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6003-94ec -- SIP/6004-4b4d is ringing -- SIP/6004-4b4d answered SIP/6003-94ec -- Stopped music on hold on SIP/6003-94ec -- Attempting native bridge of SIP/6003-94ec and SIP/6004-4b4d == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6003-94ec' -- Eybeam to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6003-8a2e, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6003-8a2e -- SIP/6005-fa6a is ringing -- SIP/6005-fa6a answered SIP/6003-8a2e -- Stopped music on hold on SIP/6003-8a2e -- Attempting native bridge of SIP/6003-8a2e and SIP/6005-fa6a == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6003-8a2e' -- 8770 to 8882 both screens are black!!! *CLI -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- 8770 to Eyebeam 8770 gets picture, Eybeam no picture -- Executing Dial(SIP/6004-5e88, SIP/6005|60|trm) in new stack -- Called 6005 -- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' -- Executing Dial(SIP/6004-2cff, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6004-2cff -- SIP/6003-322c is ringing -- SIP/6003-322c answered SIP/6004-2cff -- Stopped music on hold on SIP/6004-2cff -- Attempting native bridge of SIP/6004-2cff and SIP/6003-322c == Spawn extension (from-sip, 6003, 1) exited non-zero on 'SIP/6004-2cff' -- 8882 to Eyebeam both screens are black!!! -- Executing Dial(SIP/6005-3361, SIP/6003|60|trm) in new stack -- Called 6003 -- Started music on hold, class 'default', on SIP/6005-3361 -- SIP/6003-9ed0 is ringing -- SIP/6003-9ed0 answered SIP/6005-3361 -- Stopped music on hold on SIP/6005-3361 -- Attempting native bridge of SIP/6005-3361 and SIP/6003-9ed0 -- 8882 to 8770 8882 gets a picture -- Executing Dial(SIP/6005-abd3, SIP/6004|60|trm) in new stack -- Called 6004 -- Started music on hold, class 'default', on SIP/6005-abd3 -- SIP/6004-6381 is ringing -- SIP/6004-6381 answered SIP/6005-abd3 -- Stopped music on hold on SIP/6005-abd3 -- Attempting native bridge of SIP/6005-abd3 and SIP/6004-6381 == Spawn extension (from-sip, 6004, 1) exited non-zero on 'SIP/6005-abd3' Jul 11 15:34:27 WARNING[14974]: chan_sip.c:1046 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
RE: [Asterisk-Users] how to download chan_sip2
I believe you can find it here: http://bugs.digium.com/bug_view_page.php?bug_id=759 S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kamran Ahmad Sent: Sunday, July 10, 2005 10:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to download chan_sip2 hello http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+chan_sip2 where can i download chan_sip2.c thanks Kamran ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Xten eyeBeam Video Problems
Are you sure that the video is set up correctly? If you have a cheap webcam you have to turn off video hardware acceleration. Cheers. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Blake Krone Sent: Friday, July 08, 2005 5:53 AM To: Matt Riddell Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Xten eyeBeam Video Problems I only have the basic h.263 enabled in Xten. Everytime I start sending video it just shows noise, I can see in the log that it's trying to use the 263 codec. On 7/7/05, Matt Riddell [EMAIL PROTECTED] wrote: Blake Krone wrote: Hello all, I HAD video working before I upgraded to 1.08 (latest stable with Gentoo) and now it won't work. I just see noise bars and not the video. I know the camera works as I can use it in other programs such as AIM Yahoo. Which codec are you using for video in the eyeBeam? We have video IVR, voicemail, billing for video calls etc working fine here with multiple hardware and also the eyeBeam. My recommendation would be to allow only one video codec at a time in eyeBeam's confs. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.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/Grandstream Budgetone disconnect issue
Title: Asterisk/Grandstream Budgetone disconnect issue Might want to try updating your firmware on the Grandstream. I use this version, works great: http://gs-firmware.gratissip.dk/firmwares/1.0.6.7/ Cheers, S. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bates, Curtis Sent: Thursday, July 07, 2005 7:13 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk/Grandstream Budgetone disconnect issue I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone give me a busy signal, and does not hang up. Nothing is showing up in the messages log. Calls between the other phones work ok. Any ideas? Thanks. - A.G. Edwards Sons' outgoing and incoming e-mails are electronically archived and subject to review and/or disclosure to someone other than the recipient. - ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Simpletelecom dead?
Bruce, I too am interested in the telephone number for SimpleTelecom, as my company had put quite a large prepayment to them. You said you posted the number on this list; I searched for all post by you and did not find the posting which contained a phone number. Would you be so kind as to please re post the phone number or give me a better clue as to how to find the number you called. Very much appreciated, S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Ferrell Sent: Tuesday, July 05, 2005 3:47 PM To: C F Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Simpletelecom dead? Oh puhlease! gimme a break... Go have a look at the archives... I kinda stick out all over the place. While you're away from the computer, get your tin-foil hat adjusted... maybe adjust your meds too C F wrote: Well so for all I know you work for sipmpletelcom.com and are just trying to cover up. On 7/5/05, Bruce Ferrell [EMAIL PROTECTED] wrote: tell ya what, when everyone posts all the private backdoor numbers they have, I'll post that one C F wrote: You did send it to the list, but I'm asking you to post the phone number you used to call get a hold of someone. On 7/5/05, Bruce Ferrell [EMAIL PROTECTED] wrote: I thought I sent it out to the list when I sent it to you... I guess it didn't go C F wrote: Can you please share this with everybody? who did you speak to? on which number did you get ahold of them? ___ Asterisk-Users mailing list Asterisk-Users@lists.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] codec_speex.so not loading - fedora core 1
Hi, Today I decided to upgrade my * PBX and compiled the latest Development Head and installed it. I keep getting this message: WARNING[18268]: loader.c:313 __load_resource: libspeex.so.1: cannot open shared object file: No such file or directory Jul 5 01:26:32 WARNING[18268]: loader.c:523 load_modules: Loading module codec_speex.so failed! I searched the archives for info relating to this but no luck finding a solution other than putting noload = codec_speex.so in my /etc/asterisk/modules.conf file. Can anyone advise how to fix? Cheers, Storm. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] apps api?
I never saw much on API info; maybe it's out there somewhere? I had to hack around. It starts to make sense fast. Start by taking a look at ./asterisk/apps/app_skel.c and then look at other apps and you can start to figure out how they do things fast. Good luck, S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesse Guardiani Sent: Sunday, March 27, 2005 10:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] apps api? Hello, Is there a published apps API? Or do I need to just start reading source code? -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.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 compare with Skype
Said all that, I still would love to have a Skype channel for *. Hopefully they will release a Linux API so people can start to play (including me). S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Saturday, March 26, 2005 12:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk compare with Skype Paul Fielding wrote: You bet we have to work harder to outshine Skype. I'm all over that. But we've got bigger shoes to fill than some people realize The reverse is true, as well, as was pointed out earlier on this thread. Asterisk could do the high-quality voice if it didn't care about interoperability. Skype lacks voicemail, CDRs, queues, IVRs, conferencing, music on hold; the list goes on and on. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: direct ip-to-ip call
Hi, I use the windows client SJPhone (http://www.sjlabs.com) to make direct SIP calls. To call my * box I just enter: sip:[EMAIL PROTECTED] SIP:10.100.0.201 to call my SIP handset. Good luck, S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CuPoTKa Sent: Thursday, March 24, 2005 2:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] direct ip-to-ip call Hello! I'm searching for a way to call ATA (IAX or SIP) that is not registered with any server or proxy. Is it possible to make such a call from a softphone to an ATA just with IP? Something like (sip:// or iax://)[EMAIL PROTECTED] (where 210.12.34.45 is ATA's public ip)? Regards, CuPoTKa. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] VoIP service through Asterisk?
I've used sipphone.com before. Their rates are not the greatest. S. -Original Message- why nobody use sipphone.com to connect to asterisk ? Best Regards Zhao Zigang Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:[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] small Local telco (wifi voip) some experienceswith * ??
I've successfully implemented several VOIP over WiFi networks in the UK with excellent quality. I am currently in Canada. The trick is designing a good wifi backhaul network in quasi full duplex. Storm. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paco Perez Sent: Friday, March 18, 2005 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] small Local telco (wifi voip) some experienceswith * ?? Hello. I would like to know if somebody did a wireles voip with Asterisk PBX. I think to deploy a wireless for about 500 potential customers, it's a 3 km radius maximum coverage with houses without phone lines, I work for public places telephony small enterprises ( a common bussines in Spain) so I can get good rates from 4 telcos and do LCR at my asterisk PBX. Is anybody did this before 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Canadian Content: Telus and Shaw...
Are there any VOIP lobbyist groups in Canada? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Patterson Sent: Monday, January 17, 2005 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Canadian Content: Telus and Shaw... The CRTC is the biggest joke in the world. Ten people sit on their ass making decisions for 30+ million and no one has ever done anything to remove them from power. Heck even HBO is illegal in Canada. Why? Because they few that run the country want the many to remain their captive audience. We can go on all day about the CRTC - they represent the interests of the large companies only. Branson Patterson If they will do it, you are welcome to write the letter to CRTC and other governmental agencies for uncompetitive behavior. I think it should work. All the Best! Sergey. ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Grandstream BT100 and firmware
I put .20 on my tftp server and the BT100's uploaded fine. Before I was using .16 because .18 had some issues. We have been running .20 for a month now and no problems reported. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of My Other Email Sent: Wednesday, January 19, 2005 8:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Grandstream BT100 and firmware Here's the problem I'm having...for some reason, my BT 100 has downgraded from .18 to .16 I have downloaded the zipped file with the .20 but how now what are the instructions to actually download this to my phones? Not quite sure how to proceed. I've flashed other devices before, but usually they have a program to help you along the way. thanks, Alan ___ Asterisk-Users mailing list Asterisk-Users@lists.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] Speech to Text Conversion
This may be a bit off topic, but here is the best Speech recognition I have seen so far: VLVR -- Very Large Vocabulary Speech Recognition http://akpublic.research.att.com/projects/mohri/vlvr/ Cheers, S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Cirelle Enterprises Sent: Tuesday, November 02, 2004 4:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Speech to Text Conversion - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 01, 2004 7:50 PM Subject: Re: [Asterisk-Users] Speech to Text Conversion | Cirelle Enterprises wrote: | | has anybody found anything which works for speech to | text translation? | | Implementation being instead of (or as well as) vm wav file being sent | in email, a text translation would accompany the wav file | | | Speech to text is just around the corner. It has been for 30 years. :-) | | Steve | Ahh yes, finding which corner is the challange I suppose ... greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: doublehash patch for 1.0.1
What if you call an external system and get a voicemail. Press # to finish your message . you would have to press ##. IMHO I think most users are not sophisticated enough to transfer calls. If they are they can press ##. Or am I missing something? :) S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: Sunday, October 24, 2004 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1 I don't know but it's IMHO, this should be just the opposite. Single # for a transfer and double ## to send the key on as DTMF. How many objects in a dialplan start with a #? Lyle - Original Message - From: Randy Bush [EMAIL PROTECTED] To: Barton Hodges [EMAIL PROTECTED] Cc: splatters [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:37 AM Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1 Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: doublehash patch for 1.0.1
Personally I think like you... but I have to force myself to consider the dim wits that use my PBX. :) They are fat old men who barely understand what a telephone is... let alone VOIP. :) S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: Sunday, October 24, 2004 1:11 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1 I do some script type programing and have seen this in other uses. IMHO, it would be easier to program this way. Single # go to transfer function. Get # as first character in transfer, send out the DTMF tones instead and drop the request to transfer. I could be all wet on this, but my feeble mind sezs this makes sense from a programming perspective. Lyle - Original Message - From: Storm D. J. Petersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 3:04 PM Subject: RE: [Asterisk-Users] Re: doublehash patch for 1.0.1 What if you call an external system and get a voicemail. Press # to finish your message . you would have to press ##. IMHO I think most users are not sophisticated enough to transfer calls. If they are they can press ##. Or am I missing something? :) S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: Sunday, October 24, 2004 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: doublehash patch for 1.0.1 I don't know but it's IMHO, this should be just the opposite. Single # for a transfer and double ## to send the key on as DTMF. How many objects in a dialplan start with a #? Lyle - Original Message - From: Randy Bush [EMAIL PROTECTED] To: Barton Hodges [EMAIL PROTECTED] Cc: splatters [EMAIL PROTECTED] Sent: Sunday, October 24, 2004 11:37 AM Subject: [Asterisk-Users] Re: doublehash patch for 1.0.1 Just tried the patch you made with the latest CVS and it patches fine although it does not work. Now when I hit # it does not send the DTMF to the other side at all. Although hitting ## does get the transfer. Now # doesn't do ANYTHING :) I'm not sure why that is, it works with all our phones (Grandstream BT101s and analog phones on Grandstream ATA286s). I just tested by calling my bank's IVR. applied patch. went great. now single # does not transfer and double does. but, i am having the same problem as matthew, the # does not go through at dmtf. all other keys go through as dmtf, just not the #. this is on a spa3k. clearly * is receiving the #, as ## does do a transfer. so why is a single # not being sent onward as dtmf? randy --- ps. and i have a general wonder/question about this. is someone who uses a commercial pbx, say a meridian or whatever, unable to use ivr systems because # is not sent? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream 102 flashing
Ive had similar problems with grandstream phones. They lock up for no apparent reason and after that no matter what you do they are dead. I must of sent back 15 so far like this. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Doug Reid -Stormcorp Sent: Thursday, October 21, 2004 2:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] grandstream 102 flashing Hi Check the ports WAN and LAN they are somtimes mixed up we use alot of them. Doug Reid Director Stormcorp Network Solutions (Pty) Ltd Tel: +27 11 807 1141 Fax: +27 11 807 3504 Mobile: +27 83 989 0008 E-Mail: [EMAIL PROTECTED] Web: www.stormcorp.co.za --- NOTICE - This message contains privileged and confidential information intended only for the use of the addressee named above. If you are not the intended recipient of this message, you are hereby notified that you must not disseminate, copy or take any action in reliance on it. If you have received this message in error, please notify Stormcorp Network Solutions, its subsidiaries or associates, immediately. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Stormcorp, its subsidiaries or associates. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of BetaTeilchen Sent: Thursday, October 21, 2004 5:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream 102 flashing This flashing is an indicator for a damaged firmware in your phone. Maybe an interrupted TFTP-Download when powered up or just a wrong firmware. dean collins schrieb: Does anyone know what it means when a grandstream flashes the red key light 5 times repeatedly in cycles? I got a new handset delivered to me today, powered up fine until I tried to access it via the web interface using the password admin and then it rebooted with the lcd never displaying again and the red keys flashing 5 times then a break of 3 seconds then repeat. Cheers, Dean ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phones
Title: SIP phones Why dont you use an ATA device with a loud regular phone and/or hook up one of those really loud ringing devices you can get at a phone shop? J Just a suggestion. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Michael Di Martino Sent: Wednesday, October 20, 2004 6:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phones I am looking for a loud ringing SIP phone. I am presently using the Polycom and just cannot loud enough to hear it over the din in a collocation room. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 Over Satellite = It works !
Which Sat company are you using? I struggled and got it working on Aramiska but their latency is up to 4.5 seconds. Haven't figured out why it's so long but assume they have some bad routing issues - as they are now going to offer a new service that guarantees 600ms. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Monday, October 18, 2004 2:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IAX2 Over Satellite = It works ! How does the call feel to you? That would be a one or two second delay between your statement and the other person's reply, no? Just curious, since this could be a great solution for some of the remote areas of the US, if it is workable. I know StarBand as of two years ago would be entirely useless, but Hughes may be an alternative. -Original Message- From: Jefferson Carvalho [mailto:[EMAIL PROTECTED] Sent: Monday, October 18, 2004 2:14 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] IAX2 Over Satellite = It works ! Hello all , I´d like to annouce and share my results on using IAX2 over a satellite conn. here in Brazil. I´m using IAX2/G.729 and works very well except the latency on my link ( =~750ms ). Downstream conversation is perfect , the Upstream works fine but i feel some long delays. I don´t have any kind of echo :) I was using SIP/G.729 and i didn´t have problems ... but i have to say ... that IAX2 is better in this case. Best Regards , -Jefferson Carvalho Teresina-PI-Brazil ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Database of world area codes
Hi, I'm looking for a database with all the world's country codes and area codes. Can anyone point me into the right direction? Cheers, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why do i get this message emailed to me everytime i post?
Annoying :( S. -Original Message- From: Mail Delivery System [mailto:[EMAIL PROTECTED] Sent: Thursday, September 02, 2004 3:06 AM To: [EMAIL PROTECTED] Subject: Mail delivery failed: returning message to sender This message was created automatically by mail delivery software (HiveMail). A message that you sent could not be delivered to one of its recipients: sysad The error was: The account has reached its storage limit. -- This is a copy of the message, including all the headers. -- From [EMAIL PROTECTED] Thu Sep 02 06:05:47 2004 Received: from [69.16.138.164] (helo=lists.digium.com) by dime62.dizinc.com with esmtp (Exim 4.34) id 1C2oTT-0002ka-Iu for [EMAIL PROTECTED]; Thu, 02 Sep 2004 06:05:47 -0400 Received: from [69.16.138.164] (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 02F492FE11F; Thu, 2 Sep 2004 05:05:06 -0500 (CDT) X-Original-To: [EMAIL PROTECTED] Delivered-To: [EMAIL PROTECTED] Received: from psmtp.com (exprod5mx117.postini.com [12.158.34.89]) by lists.digium.com (Postfix) with SMTP id DBD9A2FD8A2 for [EMAIL PROTECTED]; Thu, 2 Sep 2004 05:05:02 -0500 (CDT) Received: from source ([80.242.32.2]) by exprod5mx117.postini.com ([12.158.34.245]) with SMTP; Thu, 02 Sep 2004 06:05:12 EDT Received: from ip-10-2-82-148.arc.aramiska.net (ip-213-92-130-78.aramiska-arc.aramiska.net [213.92.130.78]) by dmzms01.aramiska.net (Postfix) with ESMTP id 0674C110057; Thu, 2 Sep 2004 10:05:09 + (UTC) Received: from localhost (localhost [127.0.0.1]) by ip-10-2-82-148.arc.aramiska.net (Postfix) with ESMTP id 9D09C5D; Thu, 2 Sep 2004 10:05:06 + (UTC) Received: from laptopnt (ip-192-168-1-4.internal.petercox.aramiska.net [192.168.1.4]) by ip-10-2-82-148.arc.aramiska.net (Postfix) with SMTP id 67CE64A; Thu, 2 Sep 2004 10:05:03 + (UTC) From: Storm D. J. Petersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Jitter over Sat Date: Thu, 2 Sep 2004 03:05:02 -0700 Message-ID: [EMAIL PROTECTED] MIME-Version: 1.0 Content-Type: text/plain; charset=US-ASCII Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2910.0) X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2800.1441 In-Reply-To: [EMAIL PROTECTED] Importance: Normal X-Virus-Scanned: by Aramiska Arc X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [65/3] X-BeenThere: [EMAIL PROTECTED] X-Mailman-Version: 2.1.5 Precedence: list Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:[EMAIL PROTECTED] List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Hi, Thanks, I'll try to do a GSM Bridge call today. I understand your answer for why voicemail - as in it does not require realtime processing, but what about the echo back test? When I use echo back tests on other * servers or FWD I sound perfect - less some latency. Surely echo back is in realtime? Thanks again for your suggestion. Kind regards, S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 11:16 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Jitter over Sat On Tue, 31 Aug 2004, Storm D. J. Petersen wrote: I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - I call perfect but they cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. It sounds like you just don't have enough throughput in the one direction. Voicemail is fine because it doesn't need realtime capacty - the voice frames arriving from your side go into the captured file as they arrive, doesn't matter if your 10 second message takes 20 seconds to arrive... You could try using a lower bandwidth codec like GSM if you aren't already. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Jitter over Sat
Hello, I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - I call perfect but they cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. This is my setup: [my Grandstream]- [my * PBX]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) I've also tried: [my Grandstream]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) and: [my Grandstream]- [my * PBX]- [sat]- [friends Supra Phone] (or any other device) I've tried all combination of using SIP and IAX2 connections to bridge the calls using codecs ULAW and iLBC with all the same result. When I call my friends ECHO BACK TEST, I sound perfect (with a bit of latency). Anyone have some suggestions? Thanks kindly, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Jitter over Sat
Opps, at 3am I make stupid editing mistakes. Should read: I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - **I can hear him perfect**, but he cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. Storm D. J. Petersen mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: Tuesday, August 31, 2004 3:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Jitter over Sat Hello, I have a problem with jitter over a 2mb up 1mb down satellite connection. I call my friend over the satellite - I call perfect but they cannot make out a word I say. However if I leave him voicemail on his asterisk box, it records my voice perfect. I have this problem when calling other people as well. This is my setup: [my Grandstream]- [my * PBX]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) I've also tried: [my Grandstream]- [sat]- [friends * PBX]- [friends Supra Phone] (or any other device) and: [my Grandstream]- [my * PBX]- [sat]- [friends Supra Phone] (or any other device) I've tried all combination of using SIP and IAX2 connections to bridge the calls using codecs ULAW and iLBC with all the same result. When I call my friends ECHO BACK TEST, I sound perfect (with a bit of latency). Anyone have some suggestions? Thanks kindly, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Jitter over Sat
I don't mind latency ... it's the garbage jitter where no one can understand a word. Interestingly enough if I do this it works fine: [grandstream 1]- [sat]- [pbx in mothers house] [grandstream 2]- [sat] -/ where the grandstream phones are side by side. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard Sent: Tuesday, August 31, 2004 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Jitter over Sat On Tuesday 31 August 2004 11:36, Martin Mielke wrote: Hi there, this is just a me too... well, not exactly. I get jitter when trying to make SIP calls through Asterisk using a GPRS connection... can this be done actually? [...] Yes, we've done it over Vodaphone (I think). The lag, about 1.5s in some tests weve done, can really kill it. B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incomming call rejected using IAX2 with FWD
Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incomming call rejected using IAX2 with FWD
Hi, I cannot seem to accept incoming calls from FWD using IAX2. I followed the directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing calls fine using IAX via FWD. When someone calls me from FWD I get the following message: Chan_iax2.c:5251 socket_read: Reject connect attempt from 65.39.205.121 Any ideas? Thanks, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] where can I get toll-free number?
Does anyone have any interesting SIP service numbers like FWD 411 Tell-Me ? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Monday, June 14, 2004 4:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] where can I get toll-free number? It seems they don't want to do business at all -- From: Apache [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara via RT Sent: Monday, June 14, 2004 4:43 PM To: jay Cc: [EMAIL PROTECTED] Subject: Re: [nufone.net #1821] nufone.net Web Server Email Jay via RT wrote: I can\'t find rates or rate-centers for IAX or SIP termination on your website. Look harder. Jeremy McNamara - Thanks nufone. Not. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Monday, June 14, 2004 12:40 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] where can I get toll-free number? Good deal, I didn't know about nufone's service -- but then, how could I, it's not on their site. I've been to www.nufone.net a few times, and looking at their site, you'd think you're dealing with a front-business for the mafia. If they'd publish rates and a lists of rate-centers, they could make good business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Monday, June 14, 2004 11:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] where can I get toll-free number? On Jun 14, 2004, at 9:06 AM, Jay Milk wrote: I'm using zoneld numbers which I can terminate on any US number -- http://ld.net/mu has various options. You basically get your incoming voicepulse, broadvoice, etc line, then get an 800# to terminate on those lines and you're in asterisk. Through this, I also have tollfree numbers to my cellphones and fax... Yeah, but that's more expensive then NuFone's $0.029/minute 800 rate. If you're looking for an 800 number that points to an existing device, then ld.net probably a great way to go. If you're looking for 800 VoIP services, then there's no reason to stack services like this. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: cdr_addon_mysql.c
I got the CVR last night and compiled it. I didn't get any problems compiling. But now my uniquieid is not being populated. Any ideas? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: Sunday, June 13, 2004 12:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: cdr_addon_mysql.c In article [EMAIL PROTECTED], Caspar Arquint [EMAIL PROTECTED] wrote: Ed Devine wrote: Following the latest * CVS update, my MySQL was broken. Following the update, Asterisk-addons would compile fine, but when I ran asterisk I got the following error: ERROR[1202489024]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas I then tried using the patch (bug id 0001823) from bugs.digium.com, and found that Asterisk-addons would no longer compile, giving me the following errors: [...] I don't know, if it helps, but I also had some problems compiling cdr_addon_mysql.c, recently. To finally solve it that's what I did: 1) mkdir /tmp/A; cd /tmp/A 2) logged in to cvs 3) cvs checkout asterisk asterisk-addons 4) cd asterisk-addons 5) adjust the CFLAG section in the Makefile to look as follows: CFLAGS+=-fPIC CFLAGS+=-I../asterisk CFLAGS+=-I../asterisk/include CFLAGS+=-D_GNU_SOURCE 6) then run make You don't need to do any of the above, PROVIDED you do make install of asterisk BEFORE you try to do a make in asterisk-addons. Then the required include files have been installed in /usr/include/asterisk. I agree it's a little messy Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: cdr_addon_mysql.c
Whoops I didn't RFM... :( I forgot the CFLAGS+-DMYSQL_LOGUNIQUEID Sorry. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: Sunday, June 13, 2004 12:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: cdr_addon_mysql.c In article [EMAIL PROTECTED], Caspar Arquint [EMAIL PROTECTED] wrote: Ed Devine wrote: Following the latest * CVS update, my MySQL was broken. Following the update, Asterisk-addons would compile fine, but when I ran asterisk I got the following error: ERROR[1202489024]: cdr_addon_mysql.c:203 mysql_log: Failed to insert into databas I then tried using the patch (bug id 0001823) from bugs.digium.com, and found that Asterisk-addons would no longer compile, giving me the following errors: [...] I don't know, if it helps, but I also had some problems compiling cdr_addon_mysql.c, recently. To finally solve it that's what I did: 1) mkdir /tmp/A; cd /tmp/A 2) logged in to cvs 3) cvs checkout asterisk asterisk-addons 4) cd asterisk-addons 5) adjust the CFLAG section in the Makefile to look as follows: CFLAGS+=-fPIC CFLAGS+=-I../asterisk CFLAGS+=-I../asterisk/include CFLAGS+=-D_GNU_SOURCE 6) then run make You don't need to do any of the above, PROVIDED you do make install of asterisk BEFORE you try to do a make in asterisk-addons. Then the required include files have been installed in /usr/include/asterisk. I agree it's a little messy Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS - astman does not compile
Hi, Just a note to whoever cares that the CVS astman application does not seem to compile. I copied the /usr/src/asterisk/astman/Makefile into the CVS astman directory and it compiled and ran fine. S.
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi,
As you asked, I have included my diff to what I did for the DIAL command. I
probably didn't stick to some * pre-agreed standard of coding or something,
so if these things offend you then I suggest that you close your eyes. :)
The biggest thing to consider when you are doing a prepaid system is, what
if the person with the same account in/out calls twice? I chose, for now,
just to keep track in a database if an account is in use or not. Only
allowing calls to be placed/answered when the account was not engaged with
another call. It was that fastest way to implement my credit system. This
way is too limited for my liking and wanted to look into more on a way to
check in the scheduler to track credits used in real-time to allow multiple
calls out on the same account. I haven't had time to look into this in much
detail, but I am certain I can hack it into the Asterisk system - if not it
could always be done with an external daemon.
Let me know if anyone has thoughts about this.
Hope this helps people.
Storm.
*** app_dial.c 2004-03-18 16:09:14.0 -0800
--- ../../app_dial.c2004-06-10 17:41:14.044176497 -0700
***
*** 67,72
--- 67,73
'P[(x)]' -- privacy mode, using 'x' as database if provided.\n
'g' -- goes on in context if the destination channel hangs up\n
'A(x)' -- play an announcement to the called party, using x as
file\n
+ 'B(x)' -- Timeout in 'x' seconds after call was bridged.\n /*
CHANGE: Storm Petersen */
In addition to transferring the call, a call may be parked and then
picked\n
up by another user.\n
The optional URL will be sent to the called party if the channel
supports\n
***
*** 360,365
--- 361,367
struct localuser *u;
char info[256], *peers, *timeout, *tech, *number, *rest, *cur;
char privdb[256] = , *s;
+ char szBrdgTO[256] = , *s2;// CHANGE: buffer to store
Bridging Time out. Storm Petersen */
char announcemsg[256] = , *ann;
struct localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
***
*** 380,385
--- 382,390
struct varshead *headp, *newheadp;
struct ast_var_t *newvar;
int go_on=0;
+ time_t myt;
+ int iBrdgTO=0; /* CHANGE: Time out after call
bridged. Storm Petersen
*/
+
if (!data) {
ast_log(LOG_WARNING, Dial requires an argument
(technology1/number1technology2/number2...|optional timeout)\n);
***
*** 416,422
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
--- 421,449
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
! /*
! ** CHANGE: Added by Storm Petersen
! ** TIME OUT AFTER CALL WAS BRIDGED.
! */
! if ((s = strstr(data, B())) {
! /* Timeout after Bridging */
! strncpy(szBrdgTO, s + 2, sizeof(szBrdgTO) - 1);
! s2 = szBrdgTO;
! /* Copy the timeout string */
! while(*s2 (*s2 != ')'))
! s2++;
! if (*s2 == ')')
! {
! *s2 = '\0';
! iBrdgTO = atoi(szBrdgTO) + 1;
! }
! else {
! ast_log(LOG_WARNING, Bridge timeout lacking ')'\n);
! iBrdgTO = 0;
! }
! }
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
***
*** 703,708
--- 730,746
// Ok, done. stop autoservice
res2 = ast_autoservice_stop(chan);
}
+
+ /*
+ ** CHANGE: Added by Storm Petersen
+ ** Set TimeOut After call was Bridged.
+ */
+ if(iBrdgTO)
+ {
+ time(myt);
+ chan-whentohangup = myt + iBrdgTO;
+ }
+
res = ast_bridge_call(chan, peer, allowredir_in, allowredir_out,
allowdisconnect);
if (res != AST_PBX_NO_HANGUP_PEER)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent: Thursday, June 10, 2004 5:08 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I would be interested to share ideas, if you have guidence to offer I would
be greatful
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: 11 June 2004 00:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I found that the PREPAID system
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, Don't try to use my patch with the latest app_dial. It will only work with Release 1.0. Mine is just very clean and simple implementation to force a disconnect X seconds after a call was bridged. I was skimming though the latest source tonight and you are right, lots of nice features. I don't know if the L() option starts counting from when the call was made or after it was bridged though. Does anyone know? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of usedcanon Sent: Friday, June 11, 2004 1:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Hi, I was just going to say, the latest dial app has a lot more options, some very nice features. As for locking the account whilst in use, I do the same using a database and preventing further calls from the same account. I am actually using Pascal, to write AGI script, my requirement were simpler than whats been addressed in app_prepaid. Basically I am not usiung calling cards, rather callerid as the account. Umar. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Registration seems to timeout
Hi. Thanks for tipping me off with the new firmware. I installed it and tested the codec. Has more delay but seems to be better quality than what I was using before. Anyways, that didn't fix the SIP Registration Failure that I am getting. Any ideas? S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Richard Neese Sent: Wednesday, June 09, 2004 7:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Registration seems to timeout try changing your codec to ilbc and make sure that his gs has the latest flash to support it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I found that the PREPAID system didn't disconnect proper and tracked time from when you dialed not when your phone made connection. I ended up making my own system and had to modify the Dial app. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Umar Sear Sent: Thursday, June 10, 2004 9:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FW: question about prepaid app_prepaid Thanks to the lack of documentation, I decided to write my own AGI script (working but no where near complete) Look forward to replies and guidence on this topic. Umar. --- Yang Tao [EMAIL PROTECTED] wrote: Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GNU Licenses, Asterick, MySQL, and the Universe?
Hi, Can someone explain (or give me a link) in easy and no uncertain terms what the deal is with MySQL and Asterisk. If I run MySQL 3.x? Do I have to have a license if I sell my products? What if I sell a server with Asterisk on it? Thanks, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: question about prepaid app_prepaid
Hi,
As you asked, I have included my diff to what I did for the DIAL command. I
probably didn't stick to some * pre-agreed standard of coding or something,
so if these things offend you then I suggest that you close your eyes. :)
The biggest thing to consider when you are doing a prepaid system is, what
if the person with the same account in/out calls twice? I chose, for now,
just to keep track in a database if an account is in use or not. Only
allowing calls to be placed/answered when the account was not engaged with
another call. It was that fastest way to implement my credit system. This
way is too limited for my liking and wanted to look into more on a way to
check in the scheduler to track credits used in real-time to allow multiple
calls out on the same account. I haven't had time to look into this in much
detail, but I am certain I can hack it into the Asterisk system - if not it
could always be done with an external daemon.
Let me know if anyone has thoughts about this.
Hope this helps people.
Storm.
*** app_dial.c 2004-03-18 16:09:14.0 -0800
--- ../../app_dial.c2004-06-10 17:41:14.044176497 -0700
***
*** 67,72
--- 67,73
'P[(x)]' -- privacy mode, using 'x' as database if provided.\n
'g' -- goes on in context if the destination channel hangs up\n
'A(x)' -- play an announcement to the called party, using x as
file\n
+ 'B(x)' -- Timeout in 'x' seconds after call was bridged.\n
/* CHANGE: Storm Petersen */
In addition to transferring the call, a call may be parked and then
picked\n
up by another user.\n
The optional URL will be sent to the called party if the channel
supports\n
***
*** 360,365
--- 361,367
struct localuser *u;
char info[256], *peers, *timeout, *tech, *number, *rest, *cur;
char privdb[256] = , *s;
+ char szBrdgTO[256] = , *s2;// CHANGE: buffer to
store Bridging Time out. Storm Petersen */
char announcemsg[256] = , *ann;
struct localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
***
*** 380,385
--- 382,390
struct varshead *headp, *newheadp;
struct ast_var_t *newvar;
int go_on=0;
+ time_t myt;
+ int iBrdgTO=0; /* CHANGE: Time out after
call bridged. Storm Petersen */
+
if (!data) {
ast_log(LOG_WARNING, Dial requires an argument
(technology1/number1technology2/number2...|optional timeout)\n);
***
*** 416,422
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
--- 421,449
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
! /*
! ** CHANGE: Added by Storm Petersen
! ** TIME OUT AFTER CALL WAS BRIDGED.
! */
! if ((s = strstr(data, B())) {
! /* Timeout after Bridging */
! strncpy(szBrdgTO, s + 2, sizeof(szBrdgTO) - 1);
! s2 = szBrdgTO;
! /* Copy the timeout string */
! while(*s2 (*s2 != ')'))
! s2++;
! if (*s2 == ')')
! {
! *s2 = '\0';
! iBrdgTO = atoi(szBrdgTO) + 1;
! }
! else {
! ast_log(LOG_WARNING, Bridge timeout lacking
')'\n);
! iBrdgTO = 0;
! }
! }
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
***
*** 703,708
--- 730,746
// Ok, done. stop autoservice
res2 = ast_autoservice_stop(chan);
}
+
+ /*
+ ** CHANGE: Added by Storm Petersen
+ ** Set TimeOut After call was Bridged.
+ */
+ if(iBrdgTO)
+ {
+ time(myt);
+ chan-whentohangup = myt + iBrdgTO;
+ }
+
res = ast_bridge_call(chan, peer, allowredir_in,
allowredir_out, allowdisconnect);
if (res != AST_PBX_NO_HANGUP_PEER)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent: Thursday, June 10, 2004 5:08 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I would be interested to share ideas, if you have guidence to offer I would
be greatful
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: 11 June 2004 00:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I found that the PREPAID system didn't disconnect
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi,
As you asked, I have included my diff to what I did for the DIAL command. I
probably didn't stick to some * pre-agreed standard of coding or something,
so if these things offend you then I suggest that you close your eyes. :)
The biggest thing to consider when you are doing a prepaid system is, what
if the person with the same account in/out calls twice? I chose, for now,
just to keep track in a database if an account is in use or not. Only
allowing calls to be placed/answered when the account was not engaged with
another call. It was that fastest way to implement my credit system. This
way is too limited for my liking and wanted to look into more on a way to
check in the scheduler to track credits used in real-time to allow multiple
calls out on the same account. I haven't had time to look into this in much
detail, but I am certain I can hack it into the Asterisk system - if not it
could always be done with an external daemon.
Let me know if anyone has thoughts about this.
Hope this helps people.
Storm.
ps. Reposted this so that it would fit into the threads proper.
*** app_dial.c 2004-03-18 16:09:14.0 -0800
--- ../../app_dial.c2004-06-10 17:41:14.044176497 -0700
***
*** 67,72
--- 67,73
'P[(x)]' -- privacy mode, using 'x' as database if provided.\n
'g' -- goes on in context if the destination channel hangs up\n
'A(x)' -- play an announcement to the called party, using x as
file\n
+ 'B(x)' -- Timeout in 'x' seconds after call was bridged.\n
/* CHANGE: Storm Petersen */
In addition to transferring the call, a call may be parked and then
picked\n
up by another user.\n
The optional URL will be sent to the called party if the channel
supports\n
***
*** 360,365
--- 361,367
struct localuser *u;
char info[256], *peers, *timeout, *tech, *number, *rest, *cur;
char privdb[256] = , *s;
+ char szBrdgTO[256] = , *s2;// CHANGE: buffer to
store Bridging Time out. Storm Petersen */
char announcemsg[256] = , *ann;
struct localuser *outgoing=NULL, *tmp;
struct ast_channel *peer;
***
*** 380,385
--- 382,390
struct varshead *headp, *newheadp;
struct ast_var_t *newvar;
int go_on=0;
+ time_t myt;
+ int iBrdgTO=0; /* CHANGE: Time out after
call bridged. Storm Petersen */
+
if (!data) {
ast_log(LOG_WARNING, Dial requires an argument
(technology1/number1technology2/number2...|optional timeout)\n);
***
*** 416,422
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
--- 421,449
ast_log(LOG_WARNING, Dial argument takes format
(technology1/number1technology2/number2...|optional timeout)\n);
goto out;
}
!
! /*
! ** CHANGE: Added by Storm Petersen
! ** TIME OUT AFTER CALL WAS BRIDGED.
! */
! if ((s = strstr(data, B())) {
! /* Timeout after Bridging */
! strncpy(szBrdgTO, s + 2, sizeof(szBrdgTO) - 1);
! s2 = szBrdgTO;
! /* Copy the timeout string */
! while(*s2 (*s2 != ')'))
! s2++;
! if (*s2 == ')')
! {
! *s2 = '\0';
! iBrdgTO = atoi(szBrdgTO) + 1;
! }
! else {
! ast_log(LOG_WARNING, Bridge timeout lacking
')'\n);
! iBrdgTO = 0;
! }
! }
!
if (transfer) {
/* XXX ANNOUNCE SUPPORT */
***
*** 703,708
--- 730,746
// Ok, done. stop autoservice
res2 = ast_autoservice_stop(chan);
}
+
+ /*
+ ** CHANGE: Added by Storm Petersen
+ ** Set TimeOut After call was Bridged.
+ */
+ if(iBrdgTO)
+ {
+ time(myt);
+ chan-whentohangup = myt + iBrdgTO;
+ }
+
res = ast_bridge_call(chan, peer, allowredir_in,
allowredir_out, allowdisconnect);
if (res != AST_PBX_NO_HANGUP_PEER)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent: Thursday, June 10, 2004 5:08 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I would be interested to share ideas, if you have guidence to offer I would
be greatful
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Storm D. J.
Petersen
Sent: 11 June 2004 00:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] FW: question about prepaid
RE: [Asterisk-Users] GNU Licenses, Asterick, MySQL, and theUniverse?
Thanks! Why did * Pull the MySQL support in the current version? Storm. Storm D. J. Petersen mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, June 10, 2004 7:37 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GNU Licenses, Asterick, MySQL, and theUniverse? On Thu, 2004-06-10 at 20:53, Storm D. J. Petersen wrote: Hi, Can someone explain (or give me a link) in easy and no uncertain terms what the deal is with MySQL and Asterisk. If I run MySQL 3.x? Do I have to have a license if I sell my products? What if I sell a server with Asterisk on it? No uncertain terms would come from the license you receive the software under, easy might introduce uncertain terms. Basically, as long as you don't have G.729, you only need provide upon request the source code of asterisk. Similarly, mysql will work the same way. As soon as you stray into the parts of the code that arn't GPL you have to start licensing the code and pay for them. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GNU Licenses, Asterick, MySQL, andtheUniverse?
I have been in the archives, thanks. All the BSD vs. GNU vs. LGNU vs. licensing was giving me a headache. I was looking for like a summery I suppose on it all - not a rehashing. Closest thing I found was just a short blurb on it at www.voip-info.org, but that was about it. Thanks anyways. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, June 10, 2004 8:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] GNU Licenses, Asterick, MySQL, andtheUniverse? On Thu, 2004-06-10 at 21:49, Storm D. J. Petersen wrote: Thanks! Why did * Pull the MySQL support in the current version? Good time to browse the archive then, it is well documented and doesn't need a rehash. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I posted the modified app_dial.c and the diff file at the following URLs: http://www.dynastormtech.com/asterisk/app_dial.c http://www.dynastormtech.com/asterisk/stormp_app_dial.diff Basically it adds a B(x) option to the dial command. It will auto disconnect the call x seconds after the call was bridged (connected) - as apposed to disconnecting in x seconds after the call was dialed. Feel free to ask questions. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: Thursday, June 10, 2004 7:48 PM To: Asterisk-Users Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Importance: High Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep track in a database if an account is in use or not. Only allowing calls to be placed/answered when the account was not engaged with another call. It was that fastest way to implement my credit system. This way is too limited for my liking and wanted to look into more on a way to check in the scheduler to track credits used in real-time to allow multiple calls out on the same account. I haven't had time to look into this in much detail, but I am certain I can hack it into the Asterisk system - if not it could always be done with an external daemon. Let me know if anyone has thoughts about this. Hope this helps people. Storm. ps. Reposted this so that it would fit into the threads proper. [cut] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
I haven't looked at the CVS source yet - I will take a look and see if it's the similar or different. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Murphy Sent: Thursday, June 10, 2004 9:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Just one question about the B() option: When you say that it limits a call to X seconds, from the time the call is bridged, as opposed to from the time the call is dialed, is that comparing it to the L() option? I haven't plumbed the depths of the L() command in the current CVS source, but is this the difference you are referring to? murf [From the apps/app_dial.c source code: 'L(x[:y][:z])' -- Limit the call to 'x' ms warning when 'y' ms are left (repeated every 'z' ms)\n -- Only 'x' is required, 'y' and 'z' are optional.\n -- The following special variables are optional:\n ** LIMIT_PLAYAUDIO_CALLER(default yes) Play sounds to the caller.\n ** LIMIT_PLAYAUDIO_CALLEEPlay sounds to the callee.\n ** LIMIT_TIMEOUT_FILEFile to play when time is up.\n ** LIMIT_CONNECT_FILEFile to play when call begins.\n ** LIMIT_WARNING_FILEFile to play as warning if 'y' is defined.\n -- 'timeleft' is a special sound macro to auto-say the time left and is the default.\n\n ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: question about prepaid app_prepaid
Hi, I just looked at the CVS. It has some cool stuff in it. A lot of changes! I have looked at the ast_channel_bridge() function to see if it is any different but will take me a while to hash out. But it looks like it would be easy to implement a real time multi call credit system in it. For now my patch is really just for someone who wanted to use the Stable 1.0 Branch - rather than a development version. I personally like to stick with the latest stable release and base my code around it. Then usually I can just blame *my* code if something goes wrong. I'm going to snoop more into the CVS now. ^_^ S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Storm D. J. Petersen Sent: Thursday, June 10, 2004 9:45 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid I haven't looked at the CVS source yet - I will take a look and see if it's the similar or different. S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Murphy Sent: Thursday, June 10, 2004 9:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: question about prepaid app_prepaid Just one question about the B() option: When you say that it limits a call to X seconds, from the time the call is bridged, as opposed to from the time the call is dialed, is that comparing it to the L() option? I haven't plumbed the depths of the L() command in the current CVS source, but is this the difference you are referring to? murf [From the apps/app_dial.c source code: 'L(x[:y][:z])' -- Limit the call to 'x' ms warning when 'y' ms are left (repeated every 'z' ms)\n -- Only 'x' is required, 'y' and 'z' are optional.\n -- The following special variables are optional:\n ** LIMIT_PLAYAUDIO_CALLER(default yes) Play sounds to the caller.\n ** LIMIT_PLAYAUDIO_CALLEEPlay sounds to the callee.\n ** LIMIT_TIMEOUT_FILEFile to play when time is up.\n ** LIMIT_CONNECT_FILEFile to play when call begins.\n ** LIMIT_WARNING_FILEFile to play as warning if 'y' is defined.\n -- 'timeleft' is a special sound macro to auto-say the time left and is the default.\n\n ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration seems to timeout
Hi, I have an * server on a routable (public) IP address and a sip client behind NAT using a Grandstream phone. He is connected through a bi-directional satellite so he has a bit of latency involved. Usually I can dial this extension and them to me. But I keep getting a registration failed message. I have other sip clients not on a satellite and they dont get these time outs. So I assumed it has to do with tweaking something. Can anyone give me advice? Thanks kindly! S. NOTICE[49156]: chan_sip.c:5623 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '213.180.234.84' sip show peers Name/usernameHost Mask Port Status 5551006/5551006 213.180.234.84 (D) 255.255.255.255 5060 OK (1339 ms) ; ; SIP Configuration for Asterisk [general] dbname=asterisk ; Name of database in your Mysql server dbhost=localhost; Hostname of server dbuser=root ; Username in MySQL dbpass= ; Password for user in MySQL port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls disallow=all allow=gsm allow=ulaw allow=alaw allow=ilbc ;allow=spx ;allow=g723.1 qualify=9000; 9000 milliseconds for sip client to respond (9seconds). maxexpirey=180 ; Max length of incoming registration we allow ;defaultexpirey=1700 ; Default length of incoming/outoing [5551006] type=friend username=5551006 secret=HEY-ITS-A-SECERT host=dynamic dtmfmode=rfc2833; Choices are inband, rfc2833, or info callerid=NewPhone6 5551006 nat=yes qualify=9000; 9000 milliseconds for sipphone to respond (9seconds). ;canreinvite=no mailbox=5551006 SIP DEBUG: = Sip read: REGISTER sip:ruralsat-1.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 213.180.234.84;branch=z9hG4bK8a14e85d6ddaa98c From: 555-1006 sip:[EMAIL PROTECTED];tag=39b3b10051374e0e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 651 REGISTER Expires: 180 User-Agent: Grandstream BT100 1.0.4.54 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 213.180.234.84 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.180.234.84;branch=z9hG4bK8a14e85d6ddaa98c;received=213.180.234.84 From: 555-1006 sip:[EMAIL PROTECTED];tag=39b3b10051374e0e To: sip:[EMAIL PROTECTED];tag=as7c20ed2a Call-ID: [EMAIL PROTECTED] CSeq: 651 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 213.180.234.84:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.180.234.84;branch=z9hG4bK8a14e85d6ddaa98c;received=213.180.234.84 From: 555-1006 sip:[EMAIL PROTECTED];tag=39b3b10051374e0e To: sip:[EMAIL PROTECTED];tag=as7c20ed2a Call-ID: [EMAIL PROTECTED] CSeq: 651 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=444b39a8 Content-Length: 0 to 213.180.234.84:5060 Sip read: REGISTER sip:ruralsat-1.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 213.180.234.84;branch=z9hG4bK8a14e85d6ddaa98c From: 555-1006 sip:[EMAIL PROTECTED];tag=39b3b10051374e0e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 651 REGISTER Expires: 180 User-Agent: Grandstream BT100 1.0.4.54 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 213.180.234.84 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.180.234.84;branch=z9hG4bK8a14e85d6ddaa98c;received=213.180.234.84 From: 555-1006 sip:[EMAIL PROTECTED];tag=39b3b10051374e0e To: sip:[EMAIL PROTECTED];tag=as7c20ed2a Call-ID: [EMAIL PROTECTED] CSeq: 651 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 213.180.234.84:5060 Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.180.234.84;branch=z9hG4bK8a14e85d6ddaa98c;received=213.180.234.84 From: 555-1006 sip:[EMAIL PROTECTED];tag=39b3b10051374e0e To: sip:[EMAIL PROTECTED];tag=as7c20ed2a Call-ID: [EMAIL PROTECTED] CSeq: 651 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=444b39a8 Content-Length: 0 to 213.180.234.84:5060 Sip read: REGISTER sip:ruralsat-1.myvnc.com SIP/2.0 Via: SIP/2.0/UDP 213.180.234.84;branch=z9hG4bK6b7e7e5a73ba2ed7 From: 555-1006 sip:[EMAIL PROTECTED];tag=39b3b10051374e0e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL
[Asterisk-Users] Static Config?
Hi, I'm new to asterisk. After fiddling a bit I got it to work. It seems great. One question though, is it possible to configure asterisk when it is running? i.e. add new phones or do you have to restart it every time you want to make changes? Thanks, Storm. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
