RE: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4
I have the same challenge and issue, the server dies shortly after being fired up, although I am using Asterisk 1.2 Even with strace its very trying to work out whether the messages are errors or importance or just run of the mill All advice and options appreciated Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Thursday, 17 May 2007 11:37 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4 On Wed, May 16, 2007 at 03:22:35PM +0200, Jack wrote: Hi, has anyone managed to get hudlite server working on a Debian Etch based installation of Asterisk 1.4? So far I managed to eliminate all error messages, but the process is killed directly after starting the hudlite server without showing any error messages. I would be very happy if anyone can give me some hints or point me to a installation guide. What I would do in such a situation, is run everything under strace. However, recall that you're dealing with a proprietary program here. The only ones who have the full information to help you are Fonality. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DTMF not working using *98, but OK on inbound routes?
I have this happening with a Cisco 7960 - I can't see what the difference is, I have asterisk 1.2.13 and a number of 7960s which happily work, as well as some 7961s which also work. However one 7960 doesn't, although it dials quite happily but that's probably due to dtmf being put into SIP rather than inband. Why one works and the other doesn't I don't yet know. Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Sent: Thursday, 17 May 2007 2:40 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF not working using *98,but OK on inbound routes? Has anyone seen anything like this: I dial *98. Asterisk says Password? I punch in the password, and the system doesn't recognize the tones. However, if I dial my own number and ignore the incoming call, it goes to voicemail, and then I can get into voicemail. I have a sneaking suspicion that Asterisk is somehow not recognizing the DTMF tones somewhere along the way. This happens intermittently with Linksys ATAs and Polycom phones. Using a Cisco 3640 VOIP router. Any ideas on what to check? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call someone to instantly join conference using MeetMe
Arpit Mehta wrote on 5/19/07 10:18 PM: I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or any other conference application? Any suggestions/hints/links are welcome. Set up an extension that dials directly into the conference in question, then use that extension via the Local channel as the source of a call to the number you want to dial, triggered via the Management API or a call file. [meetme-dialin] exten = 1234,1,Answer() exten = 1234,n,MeetMe(4321) Pipe the following into the Manager API with an extra blank line at the end: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: from-inside (or whatever context is appropriate) Exten: (the number you want to call) Priority: 1 I'm going from memory, so you may have to play with it a little bit but that's the basic idea. -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call someone to instantly join conference using MeetMe
Arpit Use Auto dial. http://www.voip-info.org/wiki-Asterisk+auto-dial+out Create a .call file as mentioned by Dave. Dave Miller wrote: Arpit Mehta wrote on 5/19/07 10:18 PM: I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or any other conference application? Any suggestions/hints/links are welcome. Set up an extension that dials directly into the conference in question, then use that extension via the Local channel as the source of a call to the number you want to dial, triggered via the Management API or a call file. [meetme-dialin] exten = 1234,1,Answer() exten = 1234,n,MeetMe(4321) Pipe the following into the Manager API with an extra blank line at the end: Action: Originate Channel: Local/[EMAIL PROTECTED] Context: from-inside (or whatever context is appropriate) Exten: (the number you want to call) Priority: 1 I'm going from memory, so you may have to play with it a little bit but that's the basic idea. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dry Copper Pair
On Fri, 11 May 2007, Alex Balashov wrote: On Fri, 11 May 2007, John Treble said something to this effect: Can you still do ?homebrew? PTP T1 in the U.S. this way? I thought this was nixed by the ILEC/CLECs years ago. It's logically possible. But if you're trying to do T1 over a single pair, you'd have to break it out using HDSL/PairGain sort of line equipment, since you obviously can't install field repeaters or do any span conditioning yourself. From then on it's a crapshoot and really just depends on whether the copper is of quality, distance, specifications, etc. that can support the specification. There's no way for them to nix that, really, other than possibly keeping load coils or other constraining stuff on the facilities that tends to need to be removed for various high-speed data line / private line applications. Most telcos have long since done wholesale load coil removal sweeps (qwest did it many years ago) in preparation for dsl and other highspeed data services. -Dan___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dry Copper Pair
On Fri, 11 May 2007, Jon Pounder wrote: again, I'm interested to know anyone whose actually done this, and what the results were, since I have been thinking of the same thing for a while. Yep, did it for about 10 years straight :) both ADSL and SDSL. Most reliable service I've ever had from a telco. Our T1s, T3s, POTS etc would take a dump but our dry copper links would stay up! Never ran into load coils, just length issues. Because the connection runs from the customer to the CO and then to you. So one or both ends better be close to the CO (in this case the ISP I worked for was one block away from the CO). -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel hangs machine...
On Sat, May 19, 2007 at 07:26:33PM +0200, Angel Luis Martinez wrote: Hi all. When i do a service zaptel stop on my machine,sometimes it crash and i must unplug and plug the power cord to restart the machine. Also sometimes load zttranscode and wct4xxp, and oter times wct4xxp only... it's running centos 4.4, libpri 1.2.4, zaptel 1.2.17.1 and asterisk 1.2.18. with a TE212P. Can somebody help me? Try editing /etc/init.d/zaptel and remove 'ztcfg -s' from there. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk On Solaris 10
You would probably be better off getting support from the SolarisVoIP mailing list. Kapil Dhawan wrote: Any help is appreciated. Kapil Dhawan wrote: Hi List Whats the best way to run * on Solaris 10 with x86 architecture. I am following solarisvoip.com using svn, but came across issues like 1. app_lookupcnam compilation issue - Wrong format of ELF. Is this the correct way. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: OK to have Asterisk and clients behind firewalls?
On Fri, 18 May 2007 08:49:49 +0100 (BST), in gmane.comp.telephony.pbx.asterisk.user Gordon Henderson [EMAIL PROTECTED] wrote: Yes. You need to do a few things. Firstly, you need the asterisk server on a static IP address on the inside, so make sure it doesn't get it's IP address from the local DHCP server. Next, you need to enable port-forwarding on your router. You need to forward port 5060 and 1 through 2 to the internal IP address of your asterisk box. Finally, you need to tell the asterisk box that it's on the inside of a NAT firewall. In sip.conf, you need 3 additional lines: nat=yes localnet=192.168.4.0/24 externip=1.2.3.4 Thanks a lot :-) That solved the UNREACHABLE issue, and the remote is now ringing... ... but when I pick up the handset, I get no voice either way, even when I set the Linksys gateway to use a static external IP address (STUN doesn't seem to work). But that's another question for another thread. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [*Win32 0.60] Sending call notification by e-mail/web?
On Tue, 15 May 2007 15:52:44 -0400, in gmane.comp.telephony.pbx.asterisk.user you wrote: Freepascal seems to work very nicely. However, I'm not sure how delphi behaves with stdin/stdout since I've not written many console apps in delphi, mostly GUI rich software. The best bet would be as another poster suggested and to write a FastAGI server. Thanks everyone for the input. As *Win32 doesn't seem very reliable (crashed twice), I moved to a Linux server instead. I'll take a look at writing scripts with Delphi or Python. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to write data to astdb?
On Wed, 16 May 2007 12:17:05 +0300, in gmane.comp.telephony.pbx.asterisk.user Diego Iastrubni wrote: This will be VERY slow. Other options might be writing to the asterisk socket (I heard it's not that reliable). But again, this will be a problem on remote scenarios. What I have been using is creating a asterisk-manager connection to Asterisk, which is very reliable and fast. The downside is that you must have a user configured in manager.conf (all others do not need this, a simple root account is good enough). Thanks for the idea. I'll look into this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenWengo + Asterisk?
OpenWengo has just released WengoPhone v2.1.0: http://www.openwengo.org/index.php/openwengo/public/homePage/news?payload[newsId]=0 . Has anyone had success (or notable failures) using it as a client for Asterisk? Any advice on integrating it into dialplan, apps, config DBs, etc? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dry Copper Pair
Quoting [EMAIL PROTECTED]: On Fri, 11 May 2007, Jon Pounder wrote: again, I'm interested to know anyone whose actually done this, and what the results were, since I have been thinking of the same thing for a while. Yep, did it for about 10 years straight :) both ADSL and SDSL. Most reliable service I've ever had from a telco. Our T1s, T3s, POTS etc would take a dump but our dry copper links would stay up! Never ran into load coils, just length issues. Because the connection runs from the customer to the CO and then to you. So one or both ends better be close to the CO (in this case the ISP I worked for was one block away from the CO). how many cable feet were you ever able to actually get various speeds at ? around here it might just be the geography but I think load coils are really just a well talked about myth. There are no truly long haul lines due to the number of cities so close together and the lakes blocking what would be any longer haul lines. The trend also seems to be just drop in a chunk of fibre and park a dms switch right near any sizable new development and not feed it from the CO at all. This is something else I have been wondering about - these telco dms shelters are fed off mains power, and probably have batteries of some sort, but I am wondering how they fare in long term outages. For example our t1's are fed direct from a larger CO with generators etc. - never had a problem in the big blackout a few years ago, how long do these remotes stay up by comparison ? A related question is now with these remotes is dry copper even really physically possible even within the same city ? Another interesting observation, the cable tv utility in the same area has NG generators at every neighbourhood pop where they hop off fibre to coax (every 250 or so houses), yet on the longer wire runs they have pole mounted Alpha amplifiers fed off the utility power. Seems like the same sort of mentality, if you're not right close in, watch out if the power's off. -Dan ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID matching
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number to
check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected connect
attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]' does not
exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
___
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RE: [asterisk-users] Dry Copper Pair
On Sun, 20 May 2007, Jon Pounder said something to this effect: The trend also seems to be just drop in a chunk of fibre and park a dms switch right near any sizable new development and not feed it from the CO at all. Big Class 5 switches like DMSs do not live in remote terminals of the sort you're describing. Those are invariably CO switches, and their capacity scales far beyond serving a mere outlying area. That's like using a fire hydrant to feed a small garden hose. What they do put in remote terminals are DLCs with GR.303 trunks for backhauling customer POTS interfaces directly into a CO switch logically. DSLAMs are also often remote. These shelters are fed off mains power but typically have extensive battery plant surrounding them, and in some cases generators depending on the size of the installation and the likelyhood of an outage. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenWengo + Asterisk?
i haven't done extensive testing on the OpenWango but did get it to connect without many problems... the only issue was the software was unstable and crashed a lot (at least the Linux version was) ... now this may not have been openWango's fault... maybe it was my system. but i haven't had issues with twinkle as a softphone on Linux. On 5/20/07, Matthew Rubenstein [EMAIL PROTECTED] wrote: OpenWengo has just released WengoPhone v2.1.0: http://www.openwengo.org/index.php/openwengo/public/homePage/news?payload[newsId]=0. Has anyone had success (or notable failures) using it as a client for Asterisk? Any advice on integrating it into dialplan, apps, config DBs, etc? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
