Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
I've had this exact same problem before. if you have multiple ips make sure asterisk binds to the external ip and see if this fixes it. ___ --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] asterisk doesn't know version of asterisk-addons?
Hi! I noticed when upgrading asterisk that the latest version of asterisk is not recognizing the version of asterisk-addons properly. When you clean out /usr/lib/asterisk/modules and then install zaptel-1.2.12 - libpri-1.2.4 - asterisk 1.2.14 - asterisk-addons-1.2.5 and then you compile and install asterisk *again* it complains that the modules of asterisk-addons are not built for this version of asterisk? Weird eh? ___ --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] [OT] Wifi SIP phones - LinkSys WIP330
We tested several Wifi phone in an asterisk install: Hitachi 3000, 5000, UTstarCom F1000, and Siemens SL75WLAN. We needed roaming between 6 AP, encryption, and a reasonnable battery. The only one covering all the needs is the Siemens... And the cheaper also. Here are some conclusion: - Hitachi phones (3000 and 5000) work well with roaming, difficult to notice when the phone switch from one AP to another. But: as soon as you use WPA, not possible to be used: 3 or 4 seconds of delay!!! Batery is also a problem. - UtStarCom cannot handle roaming. - Siemens behaves very good in such environment. Roaming works well, without and with WPA. Battery duration is the best from those few phone but still not really good: standby time is around 48hrs when the phone is located close to the router, but in normal usage onfiguration (continous roaming between 6AP, a few calls - 5 to 10 calls during 5hrs), the battery is empty after 5 hrs. The siemens phone can be find in German web site, so no problem to buy it there from France, where I'm located... Hope these small lines would be helpfull for others. Alban Le Vendredi 29 Décembre 2006 08:19, Olivier a écrit : Trouble is this (promising) phone is not distributed everywhere, at least, not here in France, yet. I couldn't get any reason from Siemens France. ___ --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] vzaphfc?
On Thu, 28 Dec 2006, Gavin Hamill wrote: On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote: vzaphfc is not a complete replacement of bristuff. It replies on most of it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI driver for HFC-s-based PCI cards. Further, if you're looking for 'something else' re: cheapo ISDN cards, definately give Asterisk 1.4 and mISDN a look - no BRIStuff, no huge patches, no wacky stuff.. all Asterisk-core support that worked really well in the brief time I tested it. The key difference is rather than generating 8000 interrupts per second, the mISDN kernel driver (which itself can be thought of 'isdn4linux' version 2.0) mISDN is just one part of isdn4linux. don't forget the all other isdn drivers under linux and the core. If you talk about some 2.0 version, then you should say it is HiSax 2.0, because mISDN is the driver for the passive isdn cards like HiSax. Armin polls the card, leading to much lower system load, and no 'wanted 8 bytes, read 7!' errors from dmesg. Cheers, Gavin. ___ --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] vzaphfc?
It's not necessary to recompile the kernel for mISDN support. Check http://www.laimbock.com/asterisk/ Grab the mISDN source rpm, and build it. $ wget http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm $ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm then check /usr/src/redhat/RPMS/i386/ You should have the kernel modules and userspace applications. Once installed, I could enable chan_misdn in asterisk 1.4 without issue, and it's working great in NT mode with ISDN phones. I haven't tested asterisk 1.2, but there is no it shouldn't work as well. Julian J. M. On 12/29/06, Remco Barendse [EMAIL PROTECTED] wrote: On Thu, 28 Dec 2006, Gavin Hamill wrote: On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote: vzaphfc is not a complete replacement of bristuff. It replies on most of it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI driver for HFC-s-based PCI cards. Further, if you're looking for 'something else' re: cheapo ISDN cards, definately give Asterisk 1.4 and mISDN a look - no BRIStuff, no huge patches, no wacky stuff.. all Asterisk-core support that worked really well in the brief time I tested it. The key difference is rather than generating 8000 interrupts per second, the mISDN kernel driver (which itself can be thought of 'isdn4linux' version 2.0) polls the card, leading to much lower system load, and no 'wanted 8 bytes, read 7!' errors from dmesg. Thanks for the tip, I'll have a look at it. The main reason for me to use bristuff is that i don't want to mess mess around downloading and compiling my own kernels. I am just running CentOS 4 boxes with stock CentOS 4 kernels. Everytime I was screwing around with making my own kernels sooner or later I got bitten by screwing up the installation of the kernel and the box wouldn't boot anymore. :) On the wiki I found the manual from BeroNet which looks pretty straightforward but is for Asterisk 1.2 Any differences for Asterisk 1.4? Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vzaphfc?
On Fri, 29 Dec 2006, Julian J. M. wrote: It's not necessary to recompile the kernel for mISDN support. Check http://www.laimbock.com/asterisk/ Grab the mISDN source rpm, and build it. $ wget http://www.xs4all.nl/~pjl/downloads/asterisk/srpms/mISDN-cvs20061107-2_fc6.lc.src.rpm $ rpmbuild --rebuild mISDN-cvs20061107-2_fc6.lc.src.rpm then check /usr/src/redhat/RPMS/i386/ You should have the kernel modules and userspace applications. Once installed, I could enable chan_misdn in asterisk 1.4 without issue, and it's working great in NT mode with ISDN phones. I haven't tested asterisk 1.2, but there is no it shouldn't work as well. Julian J. M. Sounds great thanks for the pointer!! I'll give it a try tonight Cheers! Remco ___ --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] Boot load wcfxo does not configure self underUbuntu 6
On Thu, Dec 28, 2006 at 10:05:51PM -0800, Yuan LIU wrote: On Fri, Dec 15, 2006 at 06:32:19PM -0800, Yuan LIU wrote: When booting Ubuntu 6.06.1 (Linux 2.6.15-27-386), wcfxo would load but not configure. I have three ways to manually force wcfxo to configure: 1) ztcfg, 2) modprobe -f wcfxo, or of course 3) unload and reload wcfxo. Each works equally well. The usual confusion about init scripts. Debian's init scripts automatically load the module for this card using coldplug (a run of hotplug when the system starts). However the modprobe of the module fails due to the silly automatic run of ztcfg at module load tme with very stupid modprobe settings. So as a workaround you unload and reload the module. Not smart, and has a potential for races. What is the output of: grep wcfxo /etc/modprobe.d/* zaptel:install wcfxo /sbin/modprobe -s --ignore-install wcfxo $CMDLINE_OPTS /sbin/ztcfg Looks like a correct syntax. But as I explained, it better be dropped altogether. -- 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
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
On Thu, Dec 28, 2006 at 08:41:19PM -0500, Alex Robar wrote: As if we needed more proof that XXX was XX No, we don't. I have subscribed to this mailing list to take part in Asterisk-related discssion. I have not subscribed here for Psycology 101 or amature diagnostics. This thread has stopped being relevant to this list long ago. The bunch of you artificially keep it. Not to mention posting on-list messages that were sent off-list. Feel free to continue this discussion in private mail. Not in asterisk-users . Regards, -- 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] Dialed Number missing from the CDR when using call files.
The CDR, both the csv file and in MySQL does not contain the dialed number (src) in case of a call placed using .call files. ,,302,extensions,,Zap/4-1,SIP/-82c8,Dial,SIP/|40|whtWHT,2006-12-29 15:48:56,2006-12-29 15:48:59,2006- 12-29 15:50:09,73,70,ANSWERED,DOCUMENTATION *** 1. row *** calldate: 2006-12-29 15:48:56 clid: src: dst: 302 dcontext: extensions channel: Zap/4-1 dstchannel: SIP/-82c8 lastapp: Dial lastdata: SIP/|40|whtWHT duration: 73 billsec: 70 disposition: ANSWERED amaflags: 3 accountcode: uniqueid: 1167389332.202 userfield: Is this is Bug ? The cdr should have complete info, what ever the source or method of the call. -- Regards, Nasir. ___ --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-dev] Re: [asterisk-users] How accurate is show translation?
On 23 Dec 2006, at 08:47, Dinesh Nair wrote: On 12/23/06 09:51 Leo Ann Boon said the following: I would love to hear how others are using the results from show translation in system dimensioning. So far, I feel that dimensioning an Asterisk box is still mostly guesstimation :). Currently, I'm using the 30MHz per call rule to dimension. on a Pentium D 2.80Ghz, we've sustained 300 simultaneous IAX2 calls terminating in a dialplan loop that answers the call, waits 2 seconds, plays demo-instruct and loops again. a cursory examination revealed that a large portion of the CPU was used to handle NIC interrupts. occasionally we got a chan_iax2.so error which said, Maximum trunk data space exceeded to... this seems to be controlled by the MAX_TRUNKDATA constant in chan_iax2.c which is set to 40ms of SLIN for 200 calls. it'd be nice to know what this constant is for and what would the implications of increasing it be. I'm interested in this, our IAX softphone SDK is licensed by the _server_ and customers ask how many calls they can push through a server. I'm currently saying 'about 200' with decent hardware. Could you tell me which asterisk version you tested this on ? Also how many of these calls were over trunked IAX and how many 'normal' ? Thanks _very_ much. Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] one way rtp stream (Sent alwax to 127.0.0.1)
Yes, that is the solution. I have to set nat=yes in sip.conf. THX Original-Nachricht Datum: Fri, 29 Dec 2006 15:10:47 +0800 Von: Dinesh Nair [EMAIL PROTECTED] An: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1) On 12/29/06 06:04 Hans-Jürgen Brand said the following: Found problem xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't know how to change this at xlite have you tried nat=yes in sip.conf for the peer ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Music On Hold Between Servers
You've answered your own question.MoH is set on the PBX on which the user is registered. It wouldn't make sence to do it any other way. On 12/28/06, Douglas Garstang [EMAIL PROTECTED] wrote: Can someone tell me how Asterisk handles music-on-hold between servers? Documentation for this is non-existent. Lets say user A, who is registered on pbx1, calls user B, who is registered on pbx2. 1. User A puts user B on hold. The moh that is played to user B should be specified according to user A. Which pbx box should this be set on? pbx1? pbx2? Both? 2. Is the situation any different if the 'trunk' between pbx1 and pbx2 is SIP or IAX? Thanks, Doug. ___ --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] 1.4 Random disconnects
I'm experiencing the same exact issue. I haven't found a solution for it however. Jeff On Thu, 2006-12-28 at 18:28 -0500, Jason Adams wrote: Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like 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
Re: [asterisk-users] 1.4 Random disconnects
On 12/28/06, Jason Adams [EMAIL PROTECTED] wrote: Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Yes, same thing here. This seems to be the only problem we have with 1.4. We are using only SIP connections. David ___ --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] Presence issues with Got SUBSCRIBE for extensions without hint. Please add hint to s
Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten = 110,hint,SIP/110 exten = 111,hint,SIP/111 exten = 112,hint,SIP/112 exten = 113,hint,SIP/113 exten = 114,hint,SIP/114 The hints seem to be working, however why is it looking for a hint for s - should I define one? Polycom's are running 1.6.7, Asterisk is 1.2.9.1 Thanks in advance wulf ___ --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] 1.4 Random disconnects
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Friday, December 29, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 Random disconnects On 12/28/06, Jason Adams [EMAIL PROTECTED] wrote: Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Yes, same thing here. This seems to be the only problem we have with 1.4. We are using only SIP connections. David I'm wondering if this is a bug? How do I go about getting all the proper info to submit a bug? Has anyone come up with a solution? - Jason ___ --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] Presence issues with Got SUBSCRIBE for extensions without hint. Please add hint to s
Are you sure there are no VoIP Phone users with Eyebeam or even polycom requesting SUBSCRIBE for other extensions? It happened to me, that users on my network were adding Subscribe for PSTN numbers that aren't even extensions on my * server. On 12/29/06, Lorentz Hinrichsen [EMAIL PROTECTED] wrote: Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten = 110,hint,SIP/110 exten = 111,hint,SIP/111 exten = 112,hint,SIP/112 exten = 113,hint,SIP/113 exten = 114,hint,SIP/114 The hints seem to be working, however why is it looking for a hint for s - should I define one? Polycom's are running 1.6.7, Asterisk is 1.2.9.1 Thanks in advance wulf ___ --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] Realtime multiple registration for a Hard Phone Snom 360
Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can someone help for this problem ? Regards Fred ___ --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] Disconnect supervision in India?
Hey all, anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision.. Thanks -- Chris Earle System Solutions Specialist ___ --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] Error compiling chan_vpb
DiegoF wrote: hello, if somebody knows like solving this error, to him it will be been thankful. It's a build system problem; I'll look into it and try to fix it in SVN. Sorry about that. ___ --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] asterisk doesn't know version of asterisk-addons?
Remco Barendse wrote: When you clean out /usr/lib/asterisk/modules and then install zaptel-1.2.12 - libpri-1.2.4 - asterisk 1.2.14 - asterisk-addons-1.2.5 and then you compile and install asterisk *again* it complains that the modules of asterisk-addons are not built for this version of asterisk? No it does not. It tells you to _verify_ that they are built for this version of Asterisk, because it knows those modules were not installed by the Asterisk installation process itself. ___ --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] Asterisk and MiniITX setups
How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f - Mark ___ --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] Realtime multiple registration for a Hard Phone Snom 360
The device config for the Snom 360 needs to be set to adhoc mode. If you are not comfortable with hand-configuration of the extensions file, take a look at freepbx as a tool to assist you. Thanks, Bryan M. Johns Partner Shelton Johns Technology Group Office: (678) 248-2637 X: 1500 Direct: (678) 229-1809 http://www.sheltonjohns.com **Sent from my mobile phone** -Original Message- From: Frédéric Marti [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 12/29/2006 9:25 AM Subject: [asterisk-users] Realtime multiple registration for a Hard Phone Snom 360 Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can someone help for this problem ? Regards Fred ___ --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] Asterisk and MiniITX setups
On 12/29/06, Mark Greene [EMAIL PROTECTED] wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f - Mark I can get around 15-20 simultaneous SIP-2-SIP calls with no transcoding on a VIA EPIA-V 1.0 GHz, so it really depends on how many simultaneous calls you require and if there will be any transcoding involved. Regards, David ___ --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] Asterisk and MiniITX setups
Mark Greene wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f http://www.mini-box.com/s.nl/sc.8/category.15/.f Mark, I've been playing around with HP Thin Client's as of late. These units can be had quite cheaply off of ebaY. Upgrading the memory and replacing the ROM boot chip (Uses standard 44pin/44pin laptop connector) and adding a PCI riser card, would make a wonderful office PBX. This is what I've found on ebaY today: http://cgi.ebay.com/279B-HP-THIN-CLIENT-370450-002_W0QQitemZ150074535639QQihZ005QQcategoryZ11221QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] problem with VoiceMailMain
Hi all:) Can you answer,how to change parameters of VoiceMailMain application?(for example:i dont want to give permission to change password and etc) ___ --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] Asterisk and MiniITX setups
Doug, This is a great price but I have a couple questions, as I am not very savy with these systems. When you say, replace the ROM chip, what do you mean by that? Replace it with another ROM chip, and how would I go about learning to program one, etc. As for the PCI riser card, I do not see that there would be room or even a slot for a pci card in the system you linked. to. My last and most important question is, can an 800 mhz chip handle much of a load at all for asterisk once you put a PRI or FXO/S card in? I would love to pursue this. Thanks, - Mark ___ --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] PHP to call script
Using the php script below. I am able to enter my number and the number to
call, however I get the following error:
-- AGI Script cid-spoof.agi completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Executing Wait(OutgoingSpoolFailed, 4) in new stack
-- Executing BackGround(OutgoingSpoolFailed, beep) in new stack
== Spawn extension (custom-web-dial-both-called, failed, 2) exited
non-zero on 'OutgoingSpoolFailed'
-- Executing Hangup(OutgoingSpoolFailed, ) in new stack
== Spawn extension (custom-web-dial-both-called, h, 1) exited non-zero on
'OutgoingSpoolFailed'
== Manager 'admin' logged off from 127.0.0.1
Here is the PHP:
html
head
titleCall From Web!/title
/head
body
?
#--
#edit the below variable values to reflect your system/information
#--
#specify the name/ip address of your asterisk box
#if your are hosting this page on your asterisk box, then you can use
#127.0.0.1 as the host IP. Otherwise, you will need to edit the following
#line in manager.conf, under the Admin user section:
#permit=127.0.0.1/255.255.255.0
#change to:
#permit=127.0.0.1/255.255.255.0,xxx.xxx.xxx.xxx ;(the ip address of the
server this page is running on)
$strHost = 127.0.0.1;
#specify the username you want to login with (these users are defined in
/etc/asterisk/manager.conf)
#this user is the default AAH AMP user; you shouldn't need to change, if
you're using AAH.
$strUser = admin;
#specify the password for the above user
$strSecret = amp111;
-This block is not used in this script###
#specify the channel (extension) you want to receive the call requests with
#e.g. SIP/XXX, IAX2/, ZAP/, etc
#$strChannel = IAX2/XX;
#specify the context to make the outgoing call from. By default, AAH uses
from-internal
#Using from-internal will make you outgoing dialing rules apply
#$strContext = from-internal;
---End of block---###
#specify the amount of time you want to try calling the specified channel
before hangin up
$strWaitTime = 30;
#specify the priority you wish to place on making this call
$strPriority = 1;
#specify the maximum amount of retries
$strMaxRetry = 2;
#
#Should edit some things below this point to make this script work
#
#get the caller's number from the form
$strCallerNumber = $_POST['txtcallernumber'];
$strChannel = $strCallerNumber;
$strCallerId = $strCallerNumber;
#get the phone number from the posted form
$strExten = $_POST['txtphonenumber'];
$callNumber = $strExten;
#specify the caller id for the call
$strCallerId = Web will call $callNumber;
$length = strlen($strExten);
if ($length == 10 is_numeric($strExten))
{
$oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection
to host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Events: off\r\n);
fputs($oSocket, Username: $strUser\r\n);
fputs($oSocket, Secret: $strSecret\r\n\r\n);
fputs($oSocket, Action: originate\r\n);
fputs($oSocket, Channel:
SIP/Telasip-gw4/[EMAIL PROTECTED]);
fputs($oSocket, WaitTime: $strWaitTime\r\n);
fputs($oSocket, CallerId: $strCallerId\r\n);
fputs($oSocket, Exten: $callNumber\r\n);
fputs($oSocket, Context: custom-web-dial-both-called\r\n);
fputs($oSocket, Priority: 1\r\n\r\n);
fputs($oSocket, Action: Logoff\r\n\r\n);
fclose($oSocket);
?
p
table width=300 border=1 bordercolor=#0f0f0f cellpadding=3
cellspacing=0
trtd
font size=2 face=verdana,georgia
color=#00Processing call. Please wait.
/font
/td/tr
/table
?
}
else
{
?
p
table width=300 border=1 bordercolor=#0f0f0f cellpadding=3
cellspacing=0
trtd
form action=? echo $_SERVER['PHP_SELF'] ? method=post
font size=2 face=verdana,arial,georgia color=#00
Enter the number you are calling from:/font
input type=text size=20 maxlength=10 name=txtcallernumberbr
font size=2 face=verdana,arial,georgia color=#00
Enter the number where you want to call:/font
input type=text size=20 maxlength=10 name=txtphonenumberbr
input type=submit value=Make Call
/form
/td/tr
/table
/p
?
}
?
/body
/html
Here is the whats in my .confs
[custom-web-dial-both]
include = outbound-allroutes
[custom-web-dial-both-called]
exten = _.,1,Wait(4)
exten = _.,2,Background(beep)
include = outbound-allroutes
exten = h,1,Hangup
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Re: [asterisk-users] asterisk doesn't know version of asterisk-addons?
Kevin P. Fleming wrote: Remco Barendse wrote: When you clean out /usr/lib/asterisk/modules and then install zaptel-1.2.12 - libpri-1.2.4 - asterisk 1.2.14 - asterisk-addons-1.2.5 and then you compile and install asterisk *again* it complains that the modules of asterisk-addons are not built for this version of asterisk? No it does not. It tells you to _verify_ that they are built for this version of Asterisk, because it knows those modules were not installed by the Asterisk installation process itself. So exactly what is the sequence of operations required. I have read different opinions, but have not found a statement of the correct order? Download and untar zaptel, libpri,asterisk and asterisk add-ons, then compile and install zaptel, libpri and asterisk THEN asterisk add-ons, or should it be zaptel, libpri, asterisk add-ons, then finish with Asterisk? John Novack ___ --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] Binary AGI Scripts
Hi Everyone,
I'm wondering if anyone here write AGI's in compiled binaries. I'm
writing a small Cepstral AGI in Freepascal/Lazarus. I know there are
some other AGI's out there, but I wanted to add some more functionality
than what is available such as having the AGI determine if the data
argument is plain text or a path to a text file and act accordingly.
The problem that I'm having is that Asterisk is not sending back any
responses to commands. I'm using stdin/stdout through the ReadLn and
WriteLn commands in freepascal.
Reading in the initial env variables is no problem, but once I issue a
command like so:
// Create wav file from swift here
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile');
ReadLn(StringVar); // = Never returns
// Clean up code to delete file, etc
(The command STREAM FILES always comes back and complains the file is
not found when it really is there so I have settled for using the
Playback application.)
Am I mistaken in thinking that Asterisk is supposed to send back a
response over stdin?
Of course, if I do not attempt to read the response, I run into bigger
problem as the sound file will not be found because the next portion of
code deletes the file created for playback, but before asterisk has a
chance to play it!
I worked on this thing all day yesterday and tried everything that I can
think of, but this morning I figured I will ask for some help.
--
Warm Regards,
Lee
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Re: [asterisk-users] Asterisk and MiniITX setups
Mark Greene wrote: Doug, This is a great price but I have a couple questions, as I am not very savy with these systems. When you say, replace the ROM chip, what do you mean by that? Replace it with another ROM chip, and how would I go about learning to program one, etc. Instead of a hard drive in the unit, it has Windows CE burned on ROM. The chip plugs into the unit with a standard 44pin connector. You can pull the chip, but a laptop cable and plug in a laptop hard drive. From there, I have a USB DVD drive that I used to install Mandriva 2007. As for the PCI riser card, I do not see that there would be room or even a slot for a pci card in the system you linked. to. I has 1 PCI slot, HP sells a riser card and a slightly larger case cover. My last and most important question is, can an 800 mhz chip handle much of a load at all for asterisk once you put a PRI or FXO/S card in? I would love to pursue this. I haven't put it to the test, but I personally think 800mhz with 512mb of memory and a 30gb 5400rpm drive would do just fine. They do have faster models though, I've seen up to 1ghz. I have photos of my unit if you'd like me to send them to you off list. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] chan_sip loading delay in Asterisk 1.2.10
Hi, I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue. 1. chan_sip.so takes about 10 secs to load up when asterisk starts. 2. When I dialout using SIP it takes 20 secs to output -- Called SIP [EMAIL PROTECTED] and get ring back from B party... Is there any config that I can check to reduce both delays? -ag ___ --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] chan_sip loading delay in Asterisk 1.2.10
Check your DNS config. That is almost always the answer to a 10 second delay :) Steve On 12/29/06, ast guy [EMAIL PROTECTED] wrote: Hi, I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue. 1. chan_sip.so takes about 10 secs to load up when asterisk starts. 2. When I dialout using SIP it takes 20 secs to output -- Called SIP [EMAIL PROTECTED] and get ring back from B party... Is there any config that I can check to reduce both delays? ___ --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] chan_sip loading delay in Asterisk 1.2.10
ast guy wrote: Hi, I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue. 1. chan_sip.so takes about 10 secs to load up when asterisk starts. 2. When I dialout using SIP it takes 20 secs to output -- Called SIP [EMAIL PROTECTED] and get ring back from B party... Is there any config that I can check to reduce both delays? -ag ___ --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 Maybe an issue with DNS? Do you have DNS Srv allowed? Do you have a good DNS Server? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 Random disconnects
On Fri, Dec 29, 2006 at 09:12:24AM -0500, Jason Adams wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Friday, December 29, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 Random disconnects On 12/28/06, Jason Adams [EMAIL PROTECTED] wrote: Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Yes, same thing here. This seems to be the only problem we have with 1.4. We are using only SIP connections. David I'm wondering if this is a bug? How do I go about getting all the proper info to submit a bug? Has anyone come up with a solution? Solution to what? What exactly are the steps required to reproduce the problem? All I saw in this thread is some random reports of disconnects. Please enable 'full' in logger.conf and set (core) verbosity and debug to some decent value. What channels do you have configured? -- 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
Re: [asterisk-users] Asterisk and MiniITX setups
On Friday 29 December 2006 14:46, Mark Greene wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f - Mark Have you looked at AstLinux? It's perfect for small form-factor boxes. The hardware should do what you want, provided you're not doing a lot of transcoding. http://www.astlinux.org/ -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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] Asterisk and MiniITX setups
I will be doing transcoding though. ___ --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] 1.4 Random disconnects
Testing 1.4 here i got the same issue. Running tcpdump figure out that packets are sent from the sip provider or ATA to asterisk1.4 machine but asterisk doesn't reply. At really, nothing apears at /var/log/asterisk/full and looks like the sockets aren't open. After a stop now and restart all works again. I can't reproduce the bug, it occurs time to time. Not a very load server. Just 2 channels was running. -- Antonio J. S. Brandão On 12/28/06, Jason Adams [EMAIL PROTECTED] wrote: Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Thanks, Jason Jason Adams Sumo Systems 4694 Cemetery Road Suite 310 Hilliard, OH 43026 Phone | 614.433.9906 ext: 102 Fax | 614.433.9931 E-mail | [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 ___ --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] How does Sipura route incoming calls?
I have a working Asterisk 1.2 and Sipura SPA3000 combo but I would like to have more control over incoming PSTN calls to the Sipura. Right now such calls come in over the fxo port (sipurafxo1 on the PSTN tab) and are routed by dial plan 8 to the fxs port (sipurafxs1 Line 1 tab) I'm not even sure if the calls are getting to the fxs port and they sure are not going to the specified context. Does Sipura automatically route these calls? If so, to where? Is there any way to prevent Sipura from picking up the PSTN line (causing all other phones connected to the incoming PSTN line to cease ringing as if the call was picked up)? Finally, what should I put in dial plan 8 or elsewhere to send the call to a context of my choice? Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --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] TE110P with Qsig
It sounds like it isn't configured correctly. Are you sure that your cabling is ok and that your span= line is correct? Matthew Fredrickson On Dec 28, 2006, at 8:29 PM, Josué Conti wrote: Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. === == Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Dec 28 21:31:57 WARNING[5484]: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 1: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 1 Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 2: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 2 Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 3: Red Alarm Dec 28 21:31:57 WARNING[5485]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 3 Dec 28 21:31:57 WARNING[5485]: chan_zap.c:6337
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
I thought he was a PI in HIS job and did ast on the side. sorry for being a troll. it was a bad day and it affected me here. sorry guys - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 28, 2006 1:17 AM Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 ___ --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] 1.4 Random disconnects
At 03:28 PM 12/28/2006, you wrote: We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? You're not alone. Ira ___ --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] 1.4 Random disconnects
At 08:29 AM 12/29/2006, you wrote: Solution to what? What exactly are the steps required to reproduce the problem? All I saw in this thread is some random reports of disconnects. Please enable 'full' in logger.conf and set (core) verbosity and debug to some decent value. What channels do you have configured? I asked this question on the 25th but no one even noticed, he asked the same question worded differently and he got a fair number of me-to's and this response. I finally had to go back to 1.2.14 so Asterisk would stay up more than 4 or 5 calls at a time. In my case it's All SIP inside with outside split 1/4 SIP, 1/4 IAX and the rest via a Digium TDM-04. My symptoms with 1.4 were I start Asterisk, I run Asterisk -vr or whatever it was to get a console and about 4 or 5 calls later the phone system would stop working and I'd be sitting at a Linux prompt. Typing Asterisk again would allow it to run for another few calls. If I had any idea what to do to help you, I would have done it, but like many of the people here, we can make Asterisk do what we need but I've no idea what development might need to troubleshoot my problem and your blithely stating set verbosity to some decent value has absolutely no meaning to me. I promise I want to help you solve this and I know how difficult it can be to fix something that might be caused by my unexpected configuration choices or accidently missing Linux component that 1.2 didn't care about. This is the first Linux box I've ever touched, all it does is run Asterisk. I'm thankful for MC and my previous familiarity with NC so I can navigate the box without too many issues, but if you want me to help you, and I assume this goes for others in my situation, I'll need a bit more direction than you just given. Maybe this needs to be on the wikki so you can just point or so we can just search or maybe it is and you'll make me feel stupid, no matter, all that really matters to me is that the problem get fixed. I'm willing to put 1.4 up again and make calls till it crashes if I had a hope that I'd have a chance of helping you. But I need your help to get there. Ira ___ --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] Asterisk and MiniITX setups
On Fri, 29 Dec 2006, Mark Greene wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f I have a lot of installations using this board: http://www.icp-epia.co.uk/index.php?act=viewProdproductId=90 The key thing is to compile asterisk for a i586. This is vitally important, as the VIA processor on those boards is lacking some MMX instructions that asterisk uses. 20 calls aren't an issue here. Transcoding is. You *really* don't want to be using speex or ilbc here! Here is the output of a show translation recalc 30 on one of these: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 412 4 320 - -72 ulaw - 7 - 110 2 118 - -70 alaw - 7 1 -10 2 118 - -70 g726 -14 9 9 - 9 825 - -77 adpcm - 7 2 210 - 118 - -70 slin - 6 1 1 9 1 -17 - -69 lpc10 -171212201211 - - -80 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -18131321131229 - - - Gordon ___ --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] Toll free numbers
Hi,
For some reason, I seem to have issues with dailing toll free numbers
and can't seem to find out why, sometimes, I get a busy signal. Some
other times I get weird errors from the phone.
The error below was a simple busy signal.
Here's couple of my info relevant to the problem:
-- Reconfigured channel 1, PRI Signalling signalling
-- Reconfigured channel 2, PRI Signalling signalling
-- Executing Dial(SIP/107-9da02970, Zap/g1/18889554562) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/18889554562
-- Zap/1-1 is proceeding passing it to SIP/107-9da02970
-- PROGRESS with cause code 28 received
-- Zap/1-1 is making progress passing it to SIP/107-9da02970
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 918889554562, 1) exited non-zero on
'SIP/107-9da02970'
from the console I get this error Progress with cause code 28 received.
In my extensions.conf file I got this in my internal context which is
used by my sip phones:
[internal]
include = trunktollfree
include = outgoing
exten = _XXX,1,Macro(incoming,SIP/${EXTEN},${EXTEN})
exten = 200,1,Answer
exten = 200,2,MusicOnHold()
exten = 999,1,Playback(demo-echotest)
exten = 999,2,Echo
exten = 999,3,Playback(demo-echodone)
exten = 1000,1, Dial(IAX2/1000,30)
;Agent login
exten = 3001,1,AgentCallbackLogin(||[EMAIL PROTECTED])
;Agent logout
exten = 3002,1,AgentCallbackLogin(||l)
exten= 2020,1,Answer
exten= 2020,2,Ringing
exten= 2020,3,Wait(2)
exten= 2020,4,Queue(queue1)
exten= 2020,5,Hangup
It's a bit messy but it's mainly for testing.
In trunktollfree, I got this:
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten = _91800.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
zaptel.conf:
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us
And I'm using a PRI line on my server, outgoing calls are working good,
it's just my toll free calls that doesn't go through, I've probably
misconfigured something I guess...
Thanks!
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[asterisk-users] trixbox web-administration
Hi list, trixbox web-administration can be reached by host ip. since I am trying trixbox on the machine where I host my website as well, can I move trixbox main page to xxx.xxx.xxx.xxx/asterisk? which file I should move and should I modify the file? Thanks. Kurt _ Get live scores and news about your team: Add the Live.com Football Page www.live.com/?addtemplate=footballicid=T001MSN30A0701 ___ --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] Asterisk and MiniITX setups
Gordon, how did you get such good numbers? Here is my setup: [EMAIL PROTECTED]:~# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 399.054 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm pni est tm2 rng rng_en ace ace_en bogomips: 799.99 g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -111134131052 - - 202 ulaw -35 - 125 4 143 - - 193 alaw -35 1 -25 4 143 - - 193 g726 -562323 -252264 - - 214 adpcm -36 3 326 - 244 - - 194 slin -34 1 124 3 -42 - - 192 lpc10 -683535583734 - - - 226 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -69363659383577 - - - What distro are you running? On 12/29/06, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 29 Dec 2006, Mark Greene wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f I have a lot of installations using this board: http://www.icp-epia.co.uk/index.php?act=viewProdproductId=90 The key thing is to compile asterisk for a i586. This is vitally important, as the VIA processor on those boards is lacking some MMX instructions that asterisk uses. 20 calls aren't an issue here. Transcoding is. You *really* don't want to be using speex or ilbc here! Here is the output of a show translation recalc 30 on one of these: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 412 4 320 - -72 ulaw - 7 - 110 2 118 - -70 alaw - 7 1 -10 2 118 - -70 g726 -14 9 9 - 9 825 - -77 adpcm - 7 2 210 - 118 - -70 slin - 6 1 1 9 1 -17 - -69 lpc10 -171212201211 - - -80 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -18131321131229 - - - Gordon ___ --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] trixbox web-administration
Hi Kurt, You'll most likely get a better answer for this question from the Trixbox forums at Trixbox.org. Trixbox is a pretty specialized distribution of Asterisk, and this list is generally for plain vanilla asterisk-related questions. Cheers, Alex On 12/29/06, Kurt Kuo [EMAIL PROTECTED] wrote: Hi list, trixbox web-administration can be reached by host ip. since I am trying trixbox on the machine where I host my website as well, can I move trixbox main page to xxx.xxx.xxx.xxx/asterisk? which file I should move and should I modify the file? Thanks. Kurt _ Get live scores and news about your team: Add the Live.com Football Page www.live.com/?addtemplate=footballicid=T001MSN30A0701 ___ --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 -- Alex Robar [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] 1.4 Random disconnects
At 03:28 PM 12/28/2006, you wrote: We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? You're not alone. I just installed AsteriskNOW (with * 1.4) it occurs to me on call queues completelly randomly ..:o( on SIP channels.., any one have tested it using IAX2 ? jat ___ --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] Asterisk and MiniITX setups
Here are my numbers, with CentOS 4.4 processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 9 model name : VIA Nehemiah stepping: 10 cpu MHz : 533.573 cache size : 64 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr cx8 mtrr pge cmov pat mmx fxsr sse rng rng_en ace ace_en bogomips: 1067.68 g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 5 512 5 419 - -74 ulaw -16 - 1 9 2 116 - -71 alaw -16 1 - 9 2 116 - -71 g726 -22 8 8 - 8 722 - -77 adpcm -16 2 2 9 - 116 - -71 slin -15 1 1 8 1 -15 - -70 lpc10 -281414211413 - - -83 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -27131320131227 - - - On 12/29/06, C F [EMAIL PROTECTED] wrote: Gordon, how did you get such good numbers? Here is my setup: [EMAIL PROTECTED]:~# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 399.054 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm pni est tm2 rng rng_en ace ace_en bogomips: 799.99 g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -111134131052 - - 202 ulaw -35 - 125 4 143 - - 193 alaw -35 1 -25 4 143 - - 193 g726 -562323 -252264 - - 214 adpcm -36 3 326 - 244 - - 194 slin -34 1 124 3 -42 - - 192 lpc10 -683535583734 - - - 226 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -69363659383577 - - - What distro are you running? On 12/29/06, Gordon Henderson [EMAIL PROTECTED] wrote: On Fri, 29 Dec 2006, Mark Greene wrote: How well do you think asterisk could run on a miniITX board like the ones linked below with the call volume of say a small doctors office or something? http://www.mini-box.com/s.nl/sc.8/category.15/.f I have a lot of installations using this board: http://www.icp-epia.co.uk/index.php?act=viewProdproductId=90 The key thing is to compile asterisk for a i586. This is vitally important, as the VIA processor on those boards is lacking some MMX instructions that asterisk uses. 20 calls aren't an issue here. Transcoding is. You *really* don't want to be using speex or ilbc here! Here is the output of a show translation recalc 30 on one of these: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 412 4 320 - -72 ulaw - 7 - 110 2 118 - -70 alaw - 7 1 -10 2 118 - -70 g726 -14 9 9 - 9 825 - -77 adpcm - 7 2 210 - 118 - -70 slin - 6 1 1 9 1 -17 - -69 lpc10 -171212201211 - - -80 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -18131321131229 - - - Gordon ___ --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
RE: [asterisk-users] Dialed Number missing from the CDR when using callfiles.
The CDR, both the csv file and in MySQL does not contain the dialed number (src) in case of a call placed using .call files. Is this is Bug ? The cdr should have complete info, what ever the source or method of the call. I have found this same problem and have not found a solution within Asterisk. AFAIK, the CDR subsystem simply does not put the 'dialed number' in the record. Not a 'bug' so much as an unfortunate design choice. Another issue is that when an auto dial call (i.e. at .call file or manager interface 'originate' action) fails, the CDR record is cut BEFORE any dialplan entries are executed, so you can't put this information into the CDR UserField via the dialplan. The wiki implies that you can use the local channel to bypass this limitation. I've tried it, but I cannot get it to work. (I always end up with two channels bridged together when all I want is one channel dialing out to deliver a message to the called party.) The wiki stuff is here: http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels If anyone has figured out how to use the local channel to initiate an autodial out call, please respond. I'd love to see how it works. -MC ___ --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] Asterisk and MiniITX setups
Nathan, what hardware are you running? ___ --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] Asterisk and MiniITX setups
On Fri, 29 Dec 2006, C F wrote: Gordon, how did you get such good numbers? Here is my setup: [EMAIL PROTECTED]:~# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 399.054 Looks like your BIOS settings are a bit wonky... I've seen this after I accidentally shafted a board by fiddling with the video settings and had to subsequently zero the bios with the jumper... Go in and check the clock multipliers, etc. FWIW: my test machine is a very old: model name : VIA Samuel 2 cpu MHz : 533.377 cache size : 64 KB and the production machine is: model name : VIA Esther processor 1000MHz cpu MHz : 997.560 cache size : 128 KB What distro are you running? I'm running Debian sarge as the base, with a stick kernel compiled statically for the hardware. I also compile asterisk zaptel from sources then hand-craft (with the aid of some scripts!) this into a compressed initrd and put it all on a 64MB IDE flash drive which subsequently loads entirely into RAM and runs from there. (I have a 2nd 64MB flash drive for voicemail storage) Output of df -h: FilesystemSize Used Avail Use% Mounted on /dev/ram0 124M 70M 55M 57% / tmpfs 125M 0 125M 0% /dev/shm /dev/hdc2 60M 16K 60M 1% /data The board has 256MB of RAM fitted, so 128MB for the ramdisk and 128MB for run-time. Who needs fancy embedded stuff these days ;-) Gordon ___ --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] Asterisk and MiniITX setups
C F wrote: Gordon, how did you get such good numbers? model name : VIA Esther processor 1200MHz cpu MHz : 399.054 Is this accurate? A 1200mhz cpu running at 399? Under clocked? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Asterisk and MiniITX setups
C F wrote: Gordon, how did you get such good numbers? H I just noted that before I did a show translation recalc 30, I had scores as yourself: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -6464906463 185 - - 518 ulaw - 119 - 129 3 2 124 - - 457 alaw - 119 1 -29 3 2 124 - - 457 g726 - 1654949 -4948 170 - - 503 adpcm - 118 2 228 - 1 123 - - 456 slin - 117 1 127 1 - 122 - - 455 lpc10 - 1856969956968 - - - 523 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc - 405 289 289 315 289 288 410 - - - asterisk*CLI show translation recalc 30 Recalculating Codec Translation (number of sample seconds: 30) Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 5 510 5 424 - -74 ulaw -14 - 1 7 2 121 - -71 alaw -14 1 - 7 2 121 - -71 g726 -18 6 6 - 6 525 - -75 adpcm -14 2 2 7 - 121 - -71 slin -13 1 1 6 1 -20 - -70 lpc10 -241212171211 - - -81 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -23111116111030 - - - -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] 1.4 Random disconnects
On Fri, Dec 29, 2006 at 09:12:24AM -0500, Jason Adams wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Friday, December 29, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 Random disconnects On 12/28/06, Jason Adams [EMAIL PROTECTED] wrote: Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Yes, same thing here. This seems to be the only problem we have with 1.4. We are using only SIP connections. David I'm wondering if this is a bug? How do I go about getting all the proper info to submit a bug? Has anyone come up with a solution? Solution to what? What exactly are the steps required to reproduce the problem? All I saw in this thread is some random reports of disconnects. Please enable 'full' in logger.conf and set (core) verbosity and debug to some decent value. What channels do you have configured? Obviously I'm not the only one with this problem. We are using sip channels only; from our provider and internally between peers. I have set the logger.conf to 'full' and set the core verbosity and haven't noticed anything unusual so far. Although we haven't had a dropped call yet. I will continue to watch the logs and see what happens. As far as reproducing the problem it's hard to stay. I'm not sure at this point how to reproduce the issue. Sometimes it's on outbound calls (Long distance) other times it's on inbound calls. - Jason ___ --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] Asterisk and MiniITX setups
Here's mine: g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 414 4 319 - -72 ulaw -15 - 112 2 117 - -70 alaw -15 1 -12 2 117 - -70 g726 -241111 -111026 - -79 adpcm -15 2 212 - 117 - -70 slin -14 1 111 1 -16 - -69 lpc10 -271414241413 - - -82 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -26131323131228 - - - processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 9 model name : VIA Nehemiah stepping: 8 cpu MHz : 999.249 cache size : 64 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr cx8 mtrr pge cmov pat mmx fxsr sse rng rng_en ace ace_en bogomips: 2000.00 [EMAIL PROTECTED] ~]# cat /proc/meminfo MemTotal: 507928 kB MemFree: 16156 kB Buffers: 49412 kB Cached: 355180 kB SwapCached: 0 kB Active: 297232 kB Inactive: 161332 kB HighTotal: 0 kB HighFree:0 kB LowTotal: 507928 kB LowFree: 16156 kB SwapTotal: 779144 kB SwapFree: 779144 kB Dirty:1644 kB Writeback: 0 kB Mapped: 72560 kB Slab:27996 kB Committed_AS: 241580 kB PageTables: 1480 kB VmallocTotal: 507896 kB VmallocUsed: 2652 kB VmallocChunk: 504588 kB HugePages_Total: 0 HugePages_Free: 0 Hugepagesize: 4096 kB fadge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: 29 December 2006 20:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk and MiniITX setups C F wrote: Gordon, how did you get such good numbers? H I just noted that before I did a show translation recalc 30, I had scores as yourself: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - -6464906463 185 - - 518 ulaw - 119 - 129 3 2 124 - - 457 alaw - 119 1 -29 3 2 124 - - 457 g726 - 1654949 -4948 170 - - 503 adpcm - 118 2 228 - 1 123 - - 456 slin - 117 1 127 1 - 122 - - 455 lpc10 - 1856969956968 - - - 523 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc - 405 289 289 315 289 288 410 - - - asterisk*CLI show translation recalc 30 Recalculating Codec Translation (number of sample seconds: 30) Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 5 510 5 424 - -74 ulaw -14 - 1 7 2 121 - -71 alaw -14 1 - 7 2 121 - -71 g726 -18 6 6 - 6 525 - -75 adpcm -14 2 2 7 - 121 - -71 slin -13 1 1 6 1 -20 - -70 lpc10 -241212171211 - - -81 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc -23111116111030 - - - -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Asterisk and MiniITX setups
Doug Lytle wrote: C F wrote: Gordon, how did you get such good numbers? model name : VIA Esther processor 1200MHz cpu MHz : 399.054 Is this accurate? A 1200mhz cpu running at 399? Under clocked? Maybe it is intentionally underclocked to run without a cpu fan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail and ip phones
What type of phone are you using? On my Grandstream 200 under the Account tab there is an item called voicemail ID This is the extension your would call to retrieve voicemail. In my case it is extension 80, so I have just 80 entered there. when I push the messages button on the phone it immediately connects me to voicemail for the extension I am calling from. You can set it up so all extensions are on the same voicemail or grouped according to your wishes. This is done in sip.conf (mailbox=) and in voicemail.conf to define mailboxes. Incoming messages for the associated mailbox will light the mesassge waiting indicator on the phone. Doug On Fri, 29 Dec 2006, Giedrius Augys wrote: Hi, In my ip phone is voicemail indicator, and also is a voicemail button (to access to voicemail server and ant to listen voicemail). My question is how to configure this button. In configuration I need to enter URL. What is the syntax of this URL, that IP Phone could fetch this voicemail from asterisk. Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial - g option
Dial(...|30|g) does not seem to work whereas Dial(...|30|gh) works just fine __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya to Asterisk via H323
I am tasked with linking an Avaya Definity switch to an asterisk box using it's IP card that handles H.323. All my googles turn up a lot of results but nothing recent. I am able to find instructions but they are dated from 2005, and often fail halfway through. What is the best way to achieve what I want, which is two way calling between the Avaya switch and Asterisk server using h.323, and where do I need to look for setting it up on centOS 4.4? Thanks in advance, - Mark ___ --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] Need advice on dual core processing with *
I have CentOS 4.4 x86_64 running on an Pentium D 830 dual core processor with the smp kernel. Does Asterisk need to be compiled in any special way to gain performance benefits from this setup? ___ --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] Presence issues with Got SUBSCRIBE for extensions without hint. Please add hint to s
Yes, it appears that the Polycom is trying to subscribe to s - why? I've triple checked the directory xml file and it is only bw'ing 110,111,112,113,114 no other extensions. See the sip log below: -- SIP read from 192.168.1.134:5060: SUBSCRIBE sip:192.168.1.65:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bKba9b690c844C2BE1 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65 CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Max-Forwards: 70 Expires: 3600 Content-Length: 0 --- (13 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.1.134 : 5060 (NAT) Transmitting (no NAT) to 192.168.1.134:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bKba9b690c844C2BE1;received= 192.168.1.134 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65;tag=as37029a1e Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX A pbx*CLI llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:192.168.1.65:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=3b34afb0 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '113' pbx*CLI -- SIP read from 192.168.1.134:5060: SUBSCRIBE sip:192.168.1.65:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bK1a8ce17b31705644 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65 CSeq: 2 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094 Authorization: Digest username=113, realm=asterisk, nonce=3b34afb0, uri=sip:192.168.1.65:5060, response=bf28cd2382f065f3ab3502c0a98074f1, algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 --- (14 headers 0 lines)--- Found user '113' Looking for s in bella-out (domain 192.168.1.65) Scheduling destruction of call '[EMAIL PROTECTED]' in 361 ms Dec 29 08:32:32 ERROR[26486]: chan_sip.c:10988 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to s in context bella-presence Transmitting (no NAT) to 192.168.1.134:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.1.134;branch=z9hG4bK1a8ce17b31705644;received= 192.168.1.134 From: 113 sip:[EMAIL PROTECTED];tag=DAC0ED6D-27373C0E To: sip:192.168.1.65;tag=as37029a1e Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 On 12/29/06, Marco Mouta [EMAIL PROTECTED] wrote: Are you sure there are no VoIP Phone users with Eyebeam or even polycom requesting SUBSCRIBE for other extensions? It happened to me, that users on my network were adding Subscribe for PSTN numbers that aren't even extensions on my * server. On 12/29/06, Lorentz Hinrichsen [EMAIL PROTECTED] wrote: Hello all, I have a number of Polycom phones 601's and 430's and I'm seeing: Got SUBSCRIBE for extensions without hint. Please add hint to s to context local-hints in the CLI over and over. I have: [local-hints] exten = 110,hint,SIP/110 exten = 111,hint,SIP/111 exten = 112,hint,SIP/112 exten = 113,hint,SIP/113 exten = 114,hint,SIP/114 The hints seem to be working, however why is it looking for a hint for s - should I define one? Polycom's are running 1.6.7, Asterisk is 1.2.9.1 Thanks in advance wulf ___ --Bandwidth and Colocation provided by Easynews.comhttp://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 http://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] Avaya to Asterisk via H323
Mark I would start with setting up two asterisk boxes and configure an H.323 link between them, then as you have it working as you like bring the Avaya into the fold. that way you know that 50% of your settings are done (bound interfaces, settings and the like). From what I remeber Avaya may change setups from version to version. I am looking forward to tackling this on a 1000+ multi site in the neer futrure, what fun Andrew On 12/29/06, Mark Greene [EMAIL PROTECTED] wrote: I am tasked with linking an Avaya Definity switch to an asterisk box using it's IP card that handles H.323. All my googles turn up a lot of results but nothing recent. I am able to find instructions but they are dated from 2005, and often fail halfway through. What is the best way to achieve what I want, which is two way calling between the Avaya switch and Asterisk server using h.323, and where do I need to look for setting it up on centOS 4.4? Thanks in advance, - Mark ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] Need advice on dual core processing with *
- John French [EMAIL PROTECTED] wrote: I have CentOS 4.4 x86_64 running on an Pentium D 830 dual core processor with the smp kernel. Does Asterisk need to be compiled in any special way to gain performance benefits from this setup? nope -- Jason Parker Digium ___ --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] Binary AGI Scripts
use agi debug command from the Asterisk CLI to see what is going on.
Also, the last time I checked, \n is needed at the end of any
command sent to Asterisk.
Regards.
On 12/29/06, Lee Jenkins [EMAIL PROTECTED] wrote:
Hi Everyone,
I'm wondering if anyone here write AGI's in compiled binaries. I'm
writing a small Cepstral AGI in Freepascal/Lazarus. I know there are
some other AGI's out there, but I wanted to add some more functionality
than what is available such as having the AGI determine if the data
argument is plain text or a path to a text file and act accordingly.
The problem that I'm having is that Asterisk is not sending back any
responses to commands. I'm using stdin/stdout through the ReadLn and
WriteLn commands in freepascal.
Reading in the initial env variables is no problem, but once I issue a
command like so:
// Create wav file from swift here
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile');
ReadLn(StringVar); // = Never returns
// Clean up code to delete file, etc
(The command STREAM FILES always comes back and complains the file is
not found when it really is there so I have settled for using the
Playback application.)
Am I mistaken in thinking that Asterisk is supposed to send back a
response over stdin?
Of course, if I do not attempt to read the response, I run into bigger
problem as the sound file will not be found because the next portion of
code deletes the file created for playback, but before asterisk has a
chance to play it!
I worked on this thing all day yesterday and tried everything that I can
think of, but this morning I figured I will ask for some help.
--
Warm Regards,
Lee
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[asterisk-users] Re: Hi reg. 2 asterisk server
Hi Thiru - Could u tell me ,how to connect 2 asterisk server using sip as a clients... asterisk server are in same network... You can connect them either as friends or as users/peers. I generally recommend the user/peer method for connecting two servers since it clearly delineates which codecs and contexts are allowed. Your sip.conf files will look something like this: Server A sip.conf: [ToServerB] type=peer secret=fromServerAtoServerB username=fromServerAtoServerB host=ip.of.serverB qualify=1000 [FromServerB] type=user secret=fromServerBtoServerA username=fromServerBtoServerA context=extensions disallow=all allow=codecs you want to allow Server B sip.conf: [ToServerA] type=peer secret=fromServerBtoServerA username=fromServerBtoServerA host=ip.of.serverA qualify=1000 [FromServerA] type=user secret=fromServerAtoServerB username=fromServerAtoServerB context=extensions disallow=all allow=codecs you want to allow Replace the items in angle brackets xxx with your own values. Now, if you have everything loaded correctly, and you issue a sip show peers from the CLI of Server B, you should see something like: ToServerA/fromServerBtoServerA ip.of.server.A 5060 OK (37 ms) ALSO: Make sure you have the correct ports opened in both directions: 5060 TCP and UDP (this is the sip standard, but you can change it in sip.conf) 1 - 2 UDP (this is the asterisk default. You can set the exact numbers in rtp.conf) - Noah P.S. It's generally better to direct these types of questions to the entire list rather than just a few users from the list. ___ --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] Asterisk and MiniITX setups
On Fri, 29 Dec 2006, Paul wrote: Doug Lytle wrote: C F wrote: Gordon, how did you get such good numbers? model name : VIA Esther processor 1200MHz cpu MHz : 399.054 Is this accurate? A 1200mhz cpu running at 399? Under clocked? Maybe it is intentionally underclocked to run without a cpu fan. My money would be on accidental incorrect BIOS settings - eg. Load Safety defaults setting or something. I did have one of hese boards do this on me after I had to do a NV-RAM clear after screwing up the video system! These VIA processors are quite hardy - I could probably overclock them if needed - the 1GHz ones I'm using run cool to the point of being cold, as do the 533MHz ones I'm using. I have a couple of 1.2GHz ones with fans, (in a router application) and the fans are temperature controlled directly by the motherboard, and I've tried my hardest to get them to spin up when in-use and they stubbornly stay off! (they do spin up at boot time, so I know they do work! but for other applications - eg. totally fanless set-top boxes, these are the works and some of them have hardware Mpeg decode - which is OT for asterisk though!) Gordon ___ --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] TE110P with Qsig
Hi Matthew thank's will be attention. I believe that the configurations are correct, I changed of server, one another hardware and the problem remains the same. :( Changing of protocol, for euroisdn the problem remains. Stranger, does not find? Best Regards Josue zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=us defaultzone=us zapata.conf [trunkgroups] [channels] language=us context=default switchtype=qsig nsf=none pridialplan=unknown prilocaldialplan=unknown facilityenable = yes signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes restrictcid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 immediate=no callerid=asreceived musiconhold=default group=1 channel=1-15 channel=17-31 2006/12/29, Matthew Fredrickson [EMAIL PROTECTED]: It sounds like it isn't configured correctly. Are you sure that your cabling is ok and that your span= line is correct? Matthew Fredrickson On Dec 28, 2006, at 8:29 PM, Josué Conti wrote: Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. === == Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28
Re: [asterisk-users] 1.4 - G729 - Have License - No path to translatefrom Zap to IAX2
Forget about this. I rollbacked to 1.2. 1.4 features are quite useless to me without being able to use G729 codec. - Original Message - From: Aryanto Rachmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, December 28, 2006 9:58 PM Subject: [asterisk-users] 1.4 - G729 - Have License - No path to translatefrom Zap to IAX2 Hello Everybody, Since I upgraded to 1.4 I always get the difficulties as below, which I have never had in 1.2: [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Call accepted by 202.153.128.34 (format g729) [Dec 28 21:05:59] VERBOSE[1734] logger.c: -- Format for call is g729 [Dec 28 21:06:00] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 is ringing [Dec 28 21:06:00] DEBUG[1756] chan_zap.c: Requested indication 3 on channel Zap/1-1 [Dec 28 21:06:02] WARNING[1734] chan_iax2.c: Received mini frame before first full voice frame . . [Dec 28 21:06:02] WARNING[1736] chan_iax2.c: Received mini frame before first full voice frame [Dec 28 21:06:02] VERBOSE[1756] logger.c: -- IAX2/VoIPRakyat-2 answered Zap/1-1 [Dec 28 21:06:02] WARNING[1756] channel.c: No path to translate from Zap/1-1(68) to IAX2/VoIPRakyat-2(256) [Dec 28 21:06:02] WARNING[1756] app_dial.c: Had to drop call because I couldn't make Zap/1-1 compatible with IAX2/VoIPRakyat-2 I just upgraded to SVN-branch-1.4-r49020M, but doesn't help. I am using TDM400P with one FXO and one FXS. Initially I just compiled and loaded zaptel and wctdm modules. Then I tried to compile and load ztd-eth, ztd-loc, ztdummy, ztdynamic and zttranscode modules as well just to make sure, but that does not help either. I have no issue at all using any other codecs on IAX. There are some threads on this mailing list for similar issue, but mostly pointed out to G729 license. I have one as below: [Dec 28 21:02:52] VERBOSE[1440] logger.c: == G.729 Host-ID: ... [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found license 'G729-' providing 1 channels [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Found total of 1 G.729 licenses [Dec 28 21:02:52] VERBOSE[1440] logger.c: == Registered translator 'g729tolin' from format g729 to slin, cost 6 There must be something basic that I missed, maybe the new 1.4 parameters, but I don't know which ones. So please help me out. Thanks a lot in advance. Cheers, Anto ___ --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] 1.4 Random disconnects
Obviously I'm not the only one with this problem. We are using sip channels only; from our provider and internally between peers. I have set the logger.conf to 'full' and set the core verbosity and haven't noticed anything unusual so far. Although we haven't had a dropped call yet. I will continue to watch the logs and see what happens. As far as reproducing the problem it's hard to stay. I'm not sure at this point how to reproduce the issue. Sometimes it's on outbound calls (Long distance) other times it's on inbound calls. The best way to deal with a bug from an enduser standpoint: Do what you can to come up with a good description for the bug, when and how it occurs (even if randomly). If you can generate any info from the logger that's good, if not, fine. Then start a new bug report at bugs.digium.com (you have to sign up for a user account, and then use the Report Issue link), and include all this info. The key is just to include as much information as you possibly can about the bug. A bug that is this serious will probably be resolved within a very short amount of time. - Noah ___ --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] TE110P with Qsig
Hi Josué, Have you checked the strap on the TE110P board ? You must have it on the E1 position, not T1 (open ?, I don't remember at this hour, sorry). Check also without crc4. And recheck ztcfg -vvv. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Josué Conti Envoyé : vendredi 29 décembre 2006 23:27 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] TE110P with Qsig Hi Matthew thank's will be attention. I believe that the configurations are correct, I changed of server, one another hardware and the problem remains the same. :( Changing of protocol, for euroisdn the problem remains. Stranger, does not find? Best Regards Josue zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=us defaultzone=us zapata.conf [trunkgroups] [channels] language=us context=default switchtype=qsig nsf=none pridialplan=unknown prilocaldialplan=unknown facilityenable = yes signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes restrictcid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 immediate=no callerid=asreceived musiconhold=default group=1 channel=1-15 channel=17-31 2006/12/29, Matthew Fredrickson [EMAIL PROTECTED]: It sounds like it isn't configured correctly. Are you sure that your cabling is ok and that your span= line is correct? Matthew Fredrickson On Dec 28, 2006, at 8:29 PM, Josué Conti wrote: Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. === == Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:57 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of
Re: [asterisk-users] Binary AGI Scripts
Moises Silva wrote:
use agi debug command from the Asterisk CLI to see what is going on.
Also, the last time I checked, \n is needed at the end of any
command sent to Asterisk.
Regards.
Hi, sorry I have already done that, but did not mention it. The output
that is displayed when I turn agi debug on is simply the list of env.
variables being pushed out to the application and of course, the last
empty line.
After that is when my call to EXEC PLAYBACK is made and I get no response.
As for \n, I think pascal WriteLn automatically appends a newline
character, but I have tried appending it myself too like so:
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile\n'); // no work
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile' + #13); // no work
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile' + #13#10); // for SG's.
I will keep looking and trying.
Thanks for responding. Just trying to eliminate the obvious as much as
I can.
--
Warm Regards,
Lee
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Re: [asterisk-users] Toll-Free number in India
On Wed, 27 Dec 2006, Tom Lynn wrote: Can anybody point me to a vendor that can provide a toll free number that can be used in India to reach the united states? Verizon Business is telling me they can't get one. As would any other US telephone company. -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --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: RE : [asterisk-users] TE110P with Qsig
Hi Francoise, thanks will be this attention. I verified to jumper of the TE110P, That´s closed (indicating link E1), in the tests without crc4, the error is the same, I of not intervene with nothing. Stranger is that I obtain you effect normally called and received too. In the ztcfg - vv to appear the 30 channels in isdn. Thank´s in advance Best Regards Josué 2006/12/29, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hi Josué, Have you checked the strap on the TE110P board ? You must have it on the E1 position, not T1 (open ?, I don't remember at this hour, sorry). Check also without crc4. And recheck ztcfg -vvv. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Josué Conti Envoyé : vendredi 29 décembre 2006 23:27 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] TE110P with Qsig Hi Matthew thank's will be attention. I believe that the configurations are correct, I changed of server, one another hardware and the problem remains the same. :( Changing of protocol, for euroisdn the problem remains. Stranger, does not find? Best Regards Josue zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=us defaultzone=us zapata.conf [trunkgroups] [channels] language=us context=default switchtype=qsig nsf=none pridialplan=unknown prilocaldialplan=unknown facilityenable = yes signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes restrictcid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 immediate=no callerid=asreceived musiconhold=default group=1 channel=1-15 channel=17-31 2006/12/29, Matthew Fredrickson [EMAIL PROTECTED]: It sounds like it isn't configured correctly. Are you sure that your cabling is ok and that your span= line is correct? Matthew Fredrickson On Dec 28, 2006, at 8:29 PM, Josué Conti wrote: Hi all, as good? I am trying to go up a board TE110P with link E1 ISDN PRI to establish connection with a central office Siemens HiPath 4000. But I am having the following errors: Server1:~ # asterisk -r Asterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. === == Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:54 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:55 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Dec 28 21:31:56 NOTICE[5484]: chan_zap.c:8176 pri_dchannel: PRI got event: HDLC Bad FCS (8)
Re: [asterisk-users] Avaya to Asterisk via H323
Andrew, I am not so reluctant when it comes to figuring out how to link the two systems once I have asterisk working with h.323. The email was asking if someone could point me in the right direction of how to setup h.323 on asterisk. I am confident that I can handle the config from there. Do you have any thoughts on that? - Mark ___ --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] [OT] Wifi SIP phones - LinkSys WIP330
The best experience I had in using a wifi handset to connect to asterisk is a windows mobile based PDA. I had the priviledge of testing a few phones in our company to connect via VOIP. I do not have any luck using nokia E61 (doesn't register and keeps on hanging). I would think it's the same with all wifi enabled nokias. I used an Ipaq 6900 series and Asus P55 and both worked well with SIP (SJphone) and IAX (PPCIAX). For me, this would be better since I will not be carrying a phone, a PDA and a VOIP phone. It's all in one device. ___ --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] E1 controller
dear all i have 8-port e1 controller, i am some confuse about e1 commands that is when and why we use *cahnnel-group* and *pri-group* e1 controller command let me konw the above question, i have further more questions related to this issue. i shall be very thankfull to you -- With Best Regards Muhammad Aslam ul Haq ___ --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] Disconnect supervision in India?
On 12/29/06, Chris Earle [EMAIL PROTECTED] wrote: anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision.. It does not work afaik, you may not get caller id also. I tested upto 1.4b3 and no luck. raj ___ --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] Binary AGI Scripts
Lee Jenkins wrote:
Moises Silva wrote:
use agi debug command from the Asterisk CLI to see what is going on.
Also, the last time I checked, \n is needed at the end of any
command sent to Asterisk.
Regards.
Hi, sorry I have already done that, but did not mention it. The
output that is displayed when I turn agi debug on is simply the list
of env. variables being pushed out to the application and of course,
the last empty line.
After that is when my call to EXEC PLAYBACK is made and I get no
response.
As for \n, I think pascal WriteLn automatically appends a newline
character, but I have tried appending it myself too like so:
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile\n'); // no work
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile' + #13); // no work
WriteLn('EXEC PLAYBACK /tmp/NewlyCreatedFile' + #13#10); // for SG's.
Have you tried using the agi unit at
http://home.cogeco.ca/~camstuff/agiunitpas.txt?
Leo
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