Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote: On Thu, May 10, 2007 23:44, Gordon Henderson wrote: On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote: It gives me pause though... Maybe it's time to get rid of my fixed line... ;-) No ;-) needed - I have friends on cable internet with no separate copper phone line now. I'd consider it myself if I weren't tied to having ADSL over my phone line, and as yet there isn't a way to separate them (in the UK) In NL there is... ;-) Especially interesting as I have ISDN, which is almost twice as expensive... So I am really going to look in to it... I'd save about EUR 20,00 per month that way! If you think your ISP is reliable enough then go for it! There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. It can work out a lot cheaper than going down the traditional ISDN2/ISDN30 route for a lot of people as a small business expands. Undfortunately I'll have to pay reconnection fee before I can cancel! :-o I guess that's a country thing - good luck :) 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] Any other softPBX like Asterisk?
Francesco Peeters (Asterisk) wrote: On Fri, May 11, 2007 07:34, Armin Schindler wrote: On Thu, 10 May 2007, Crazy Boy wrote: Hi Friends, Can anybody tell me other softPBX softwares like Asterisk? - OpenPBX - Freeswitch Or try Googling for something like 'open source pbx'... Sheesh! :-o They call him Crazy Boy for a reason. -Stephen- ___ --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] record voice
Hello everybody! I have a problem recording voices for my Asterisk menu. I used the Record(/home/lazkano/bienvenido:gsm) function to record the menu voices, but when I call from outside or from an extension the voice listen so low. is there any software to record my voice properly and convert to gsm format? Someone use an other function for that? Thank a lot to everybody. Enjoy your weekend!!!___ --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 downside of Asterisk and least cost routing...
There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. Indeed, many of our clients are doing just that. I would, however, strongly recommend against ditching PSTN entirely (in the UK, it's virtually impossible anyway since ADSL requires a PSTN line over which to run) - those PSTN lines are still useful for things like emergency service calls, directory enquiries, etc. etc. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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] Cutted audio or 2/3s blanks on EuroISDN - Asterisk1.4
Steve Totaro a écrit : Hi Steve Your Zap conf files would be helpful. Zttest results? Cat /proc/interrupts. Sharing interrupts? No. Zap con files should not be relevant as we are using ISDN. [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/zaptel.conf loadzone = us defaultzone=us [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /etc/asterisk/zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; ; You need to restart Asterisk to re-configure the Zap channel ; CLI reload chan_zap.so ; will reload the configuration file, ; but not all configuration options are ; re-configured during a reload. [trunkgroups] [channels] ; context=default ; switchtype=national ; signalling=fxo_ls ; rxwink=300 ; Atlas seems to use long (250ms) winks ; usecallerid=yes ; hidecallerid=no ; callwaiting=yes ; usecallingpres=yes ; callwaitingcallerid=yes ; threewaycalling=yes ; transfer=yes ; canpark=yes ; cancallforward=yes ; callreturn=yes ; echocancel=yes ; echocancelwhenbridged=yes ; rxgain=0.0 txgain=0.0 ; group=1 ; make these both the same. Groups range from 0 to 63. ; callgroup=1 pickupgroup=1 ; immediate=no [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 109917508 0 0 0IO-APIC-edge timer 1: 12365 0 0 0IO-APIC-edge i8042 8: 444560118 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 12: 11367 0 0 0IO-APIC-edge i8042 14:3944731 0 0 0IO-APIC-edge ide0 58: 0 0 0 0 IO-APIC-level uhci_hcd:usb1, uhci_hcd:usb3, ehci_hcd:usb5 66: 0 0 0 0 IO-APIC-level uhci_hcd:usb2, uhci_hcd:usb4 74:4552211 0 0 0 IO-APIC-level libata 90: 18418187 0 0 0 PCI-MSI eth0 98: 27358592 0 0 0 IO-APIC-level HFC-multi 106: 27358571 0 0 0 IO-APIC-level HFC-multi NMI: 14333691827 1273 LOC: 109917988 109917975 109917950 109917910 ERR: 0 MIS: 0 We use ztdummy for Meetme: [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ sudo ./zttest Opened pseudo zap interface, measuring accuracy... 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.951172% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.951172% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% --- Results after 87 passes --- Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952721 lsmod, zttranscode was loaded, I remove it: [EMAIL PROTECTED]:/home/asterisk/1.4/zaptel$ lsmod Module Size Used by ztdummy10056 0 tcp_diag6400 0 inet_diag 16784 1 tcp_diag mISDN_dsp 201384 1 hfcmulti 79884 1 mISDN_capi107116 1 l3udss146744 1 mISDN_l2 44616 1 mISDN_l1 17560 1 mISDN_core 88224 6 mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 capi 23616 0 capifs 11152 2 capi kernelcapi 56640 2 mISDN_capi,capi zaptel197608 7 ztdummy crc_ccitt 6784 1 zaptel ipv6 285664 34 ppdev 14088 0 parport_pc 41640 0 lp 17736 0 parport44684 3 ppdev,parport_pc,lp button 12192 0 ac 10376 0 battery15496 0 dm_snapshot20664 0 dm_mirror 25216 0 dm_mod 62800 2 dm_snapshot,dm_mirror loop 20112 0 tsdev 13056 0 i2c_i801 13076 0 serio_raw 12036 0 i2c_core 27776 1 i2c_i801 pcspkr 7808 0 psmouse44432 0 shpchp 42156 0 pci_hotplug20872 1 shpchp evdev 15360 1 ext3
[asterisk-users] Reminder: Asterisk Users Conference Friday 12:30PM EDT
Friday May 11, 2007 at 12:30PM EDT A short reminder that you can connect with others in the asterisk community by phone or SIP (or both obviously) during these conferences. Anyone interested in asterisk is welcome to join the conference. Details are found here: http://x2z.eu Past recordings are here: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 If we're nice to the Digium guys, they may be there as they often are. ___ --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] record voice
thankyou very much, i will probe it byee - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 10:35 AM Subject: Re: [asterisk-users] record voice On Fri, 11 May 2007, Josu Lazkano Lete wrote: Hello everybody! I have a problem recording voices for my Asterisk menu. I used the Record(/home/lazkano/bienvenido:gsm) function to record the menu voices, but when I call from outside or from an extension the voice listen so low. is there any software to record my voice properly and convert to gsm format? Someone use an other function for that? Audacity can record sound from a PC's microphone, (or better, the line-in socket if you have a good pre-amp and proper microphone) manipulate it, etc. It's also cross platform (Win/Linux/Mac) http://audacity.sourceforge.net/ You could then store your prompts in all the codec formats you support, then asterisk wouldn't have to do transcoding either. 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] record voice
On Fri, 11 May 2007, Josu Lazkano Lete wrote: Hello everybody! I have a problem recording voices for my Asterisk menu. I used the Record(/home/lazkano/bienvenido:gsm) function to record the menu voices, but when I call from outside or from an extension the voice listen so low. is there any software to record my voice properly and convert to gsm format? Someone use an other function for that? Audacity can record sound from a PC's microphone, (or better, the line-in socket if you have a good pre-amp and proper microphone) manipulate it, etc. It's also cross platform (Win/Linux/Mac) http://audacity.sourceforge.net/ You could then store your prompts in all the codec formats you support, then asterisk wouldn't have to do transcoding either. 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] Polycom power over ethernet (PoE) cables for 500/501, 600/601 and 650 sets
Stephen Bosch wrote: 3. I thought I might save some clutter by putting these cables between the midspan and the patch panel, but then I discovered that the male end of the cable is keyed, just as in the default AC cables provided with the phones, meaning that they'll only work if plugged directly into the phone itself. The reduction in clutter with this set-up is, unfortunately, not what I had hoped, though anything is better than nothing. I imagine it would work if I sanded away the plastic post on the connector, but that says nothing about how it might behave if a non-compliant device were plugged into it. Better safe than sorry. Actually, I did exactly this with the default AC cables. I plugged my Polycom wall warts into my UPS near my household patch panel, filed off the tabs on the AC cables and used them as patch cables between my (non POE) switch and the patch panel. I use a standard patch cable for the phones. I don't think this would be a good idea for an office environment. I'm not sure what would happen if I plugged something other than one of my Polycom phones (501's) into the non standard powered ethernet jack. I'm fairly safe in my home environment, since I am usually the only one messing with ethernet cables in the house, and I have told my family specifically not to ever unplug one of the phones in order to plug in something else (they can always plug into the back of the phone instead if they need a temporary connection). John ___ --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] Any other softPBX like Asterisk?
2007/5/11, Stephen Bosch [EMAIL PROTECTED]: Francesco Peeters (Asterisk) wrote: On Fri, May 11, 2007 07:34, Armin Schindler wrote: On Thu, 10 May 2007, Crazy Boy wrote: Hi Friends, Can anybody tell me other softPBX softwares like Asterisk? - OpenPBX - Freeswitch Or try Googling for something like 'open source pbx'... Sheesh! :-o They call him Crazy Boy for a reason. -Stephen- ___ --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 Other: sipX http://www.sipfoundry.org/features.html roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar ___ --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] Log CODECS in CDR's
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729 Dial then negotiates GSM for the outbound
leg,
I want to log both these codecs in a CDR.
At the moment to find the codecs used I have to look though the sip
trace or show channels/show channel (annoying when you have 50+
channels).
Im just trying to find an easier and quicker way to keep track of the
codecs used to help with debug etc.
The closest variable iv found is, ${SIP_CODEC} Set the SIP codec for a
call
Ill see if NoOp (${SIP_CODEC}) shows the codec that was used without me
setting it though I don't think it will.
Iv looked all over and I cant find anything so it looks like I may have
to hack a ast_set_var into app_dial or chan_sip
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dave
cantera
Sent: 11 May 2007 03:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Log CODECS in CDR's
morgan,
I've seen some info on additional variables in the CDR... but haven't
tried it... look to these pages:
daveC
http://www.asterisk.org/doxygen/1.2/AstCDR.html
In addition, you can set your own extra variables by using
Set(CDR(name)=value).
These variables can be output into a text-format CDR by using the
cdr_custom
CDR driver; see the cdr_custom.conf.sample file in the configs directory
for
an example of how to do this.
-and-
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Morgan Gilroy wrote:
Hi,
Does anyone know how to get the codec that was negotiated for a call
after a dial? I want to log them into CDR but can't find any way to do
it without hacking the code.
It would be good if I could get it in an asterisk variable I can log
off
seperatly.
Thanks!
___
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[asterisk-users] A couple of questions for the Mitel gurus (phone-related - not systems)
Hi Folks, Just in case there are any Mitel gurus here: 1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the SIP firmware? I have inherited one that's Minet only. 2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's lost the connecting lead. Can anyone recommend anywhere in the UK for a replacement lead or confirm the pin-out so I can check whether a generic RJ-RJ lead will work without frying anything. Thanks Nigel Kendrick ___ --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] Dundi and unknown remote peers
Hi guys, Is it possible to allow remote peers to connect to your local DUNDi Asterisk box, even if you don't have them listed in the dundi.conf? Alex ___ --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] Rapid DTMF missing digits
Version 1.4.2 but to be honest I have no reason at all to suspect that this is a problem with the asterisk software. I've able to replicate this from a few different client net connections and a across a few different linksys ata's. Where when you call into the host and enter the extension to connect to you miss the last digit of the extension. Almost every time you miss the last digit of the extension (in a 4 digit extension). My suspicion is simply because of the network we are currently using to host the asterisk box, as a packet dump on the lan segment clearly showed that the ATA transmitted all digits (rfc2833) but the asterisk host only recieved 3 of the 4. The second you dial slower everything works fine; also the lines for voice are clear with no noticeable impairments. I'm more curious if anyone else has ever run into a similar problem and what the resolution was if they found one (IE a sturdier net connection for the asterisk host), or Tweaking the timers on the ata's to slow down how fast and how long they transmit digits. I've done a few different tests and if I use a 'softphone' dialing directly into the server things work perfectly. I can dial as fast as I want, however when I come in through the pstn trunks through the upstream provider I find this problem. has anyone else ever seen this? Or seen a case where mis-matched dtmf modes across multiple providers causes this problem? minor detail on what I referred to as the 'pstn trunks' I have no analog or digital circuts all handoffs are sip. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird Saving Lost Packets since 1994 Have you seen this packet? 101010010 ___ --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] Any other softPBX like Asterisk?
On Fri, 2007-05-11 at 07:33 -0300, Roberto Pereyra wrote: - OpenPBX - Freeswitch Other: sipX Yet another: Yate http://yate.null.ro/pmwiki/ ciao Luca ___ --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] 'Invalid characters in name' with asterisk-gui
Hi all! Is there a way to asterisk-gui to allow underline (as such cpd_tom) in Names? It allows to [di]enable alphanumeric, but not underline noway. Why such restriction in asterisk-gui if even asterisk users.conf allows (and works fine) it? Thank you, Tom Lobato ___ --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 a RTP/SIP Proxy to be used as SBC (Session Border Controller)
Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM ___ --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] Confference function
- Original Message - From: Ed Nuñez To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, April 30, 2007 1:36 PM Subject: [asterisk-users] Confference function I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. My problem right now is being able to disconnect the third party and keeping the caller on the line. Would this be a function of Asterisk or the SIP / IAX phone? Any comments would be appreciated. Thank you Ed Nuñez The following page may help you with this: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ___ --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] TDM410P
Great !!! Thanks a lot !! Nitesh Divecha a écrit : Hello, Here is my config: - /etc/zaptel.conf # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 /etc/asterisk/zapata-channels.conf You need to #include zapata-channels.conf in your zapata.conf ; signalling = pri_cpe is USER ; signalling = pri_net is NETWORK group = 1 switchtype = national signalling = pri_net context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_net context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_net context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 I use FreePBX as my front-end to route calls... so I just assign the trunk groups which I want to use... Regards, Nitesh Alexandre VERNIOL wrote: HI all, Does some one can give me his configuration (zapta.conf, zaptel.conf, sample for extensions.conf) for TDM410P card or similar (1/2/3/4 BRI card) Thanks in advance. Cheers, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Module wctdm24xxp not found - TDM808P on debian
I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp i get this error FATAL: Module wctdm24xxp not found. FATAL: Error running install command for wctdm24xxp I think i have successfully compiled the zaptel drivers, and the card appears when i do a lspci 02:06.0 Ethernet controller: Digium, Inc. Unknown device 0800 (rev 11) Subsystem: Digium, Inc. Unknown device 0800 Flags: bus master, medium devsel, latency 64, IRQ 3 I/O ports at e800 [size=256] Memory at fe20 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 2000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 I've searched for a solution with no success. Another problem is that when i do a ztcfg -vv i get Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected I don't know if the errors are connected and would appreciate some help. Thanks in advance -- Juliano F. Schroeder -- Solucionathica http://www.solucionathica.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
Re: [asterisk-users] Need some help with a very simple Queestion..
Hi Gavin, You don't need queues to ring two phones, you can simply use the dial command: Dial(SIP/1SIP/10001) -- Would dial SIP extensions 1 and 10001. Now if you want the ability to have multiple people waiting on the line for those two extensions, that's when you need to look at the option of queues. Cheers, AR On 5/11/07, Gavin Spurgeon [EMAIL PROTECTED] wrote: Hi List, Just a simple question for the list this time.. I need to setup 2 Phones than can Both ring when an incoming call is made to a certain number... I have done this 3,000,000s times with CCM and have no problems with it, But it is the 1st time I have needed to do this with Asterisk. I think it can be done using Queues/Agents but I'm just unsure how do it.. The setup in question is a very small 5 Phones System based on SME 7.1 SAIL (Asterisk web interface) I have a small Sandbox setup here with me to test the test before I need to go set it up on the live system. My test phone is a Grandstream GXP 2000 but I will be using SPA-941's in the Live Environment Any help with this simple question would be great. Best Regards Gavin Spurgeon Systems Administrator Leigh City Technology College [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 -- 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
[asterisk-users] Strange problem with asterisk
Situation such. There is an asterisk working as office pbx. 6 fxo - 18 fxs ports. All works perfectly, but some times in a week something occurs. Could not catch what exactly yet. But symptoms such. The asterisk infinitely writes the message of a type to broad gullies: WARNING [20757] chan_zap.c: We're Zap/8-1, not ... ZOMBIE. Numbers of channels can change. Because of that that broad gullies get littered fairly promptly, I have not time to see that occured in an instant of the beginning of this event. When the asterisk is in such condition, the appropriating channel does not work, in this case 8. What can it be? asterisk version 1.2.14-BRIstuffed-0.3.0-PRE-1x ___ --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] Log CODECS in CDR's
On 5/11/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
At the moment to find the codecs used I have to look though the sip
trace or show channels/show channel (annoying when you have 50+
channels).
Im just trying to find an easier and quicker way to keep track of the
codecs used to help with debug etc.
The closest variable iv found is, ${SIP_CODEC} Set the SIP codec for a
call
Ill see if NoOp (${SIP_CODEC}) shows the codec that was used without me
setting it though I don't think it will.
Iv looked all over and I cant find anything so it looks like I may have
to hack a ast_set_var into app_dial or chan_sip
1.4 has the CHANNEL function:
pbxlab-01*CLI show function CHANNEL
pbxlab-01*CLI
-= Info about function 'CHANNEL' =-
[Syntax]
CHANNEL(item)
[Synopsis]
Gets/sets various pieces of information about the channel.
[Description]
Gets/set various pieces of information about the channel.
Standard items (provided by all channel technologies) are:
R/O audioreadformatformat currently being read
R/O audionativeformat format used natively for audio
R/O audiowriteformat format currently being written
R/W callgroup call groups for call pickup
R/O channeltypetechnology used for channel
R/W language language for sounds played
R/W musicclass class (from musiconhold.conf) for hold music
R/W rxgain set rxgain level on channel drivers that support
it
R/O state state for channel
R/W tonezone zone for indications played
R/W txgain set txgain level on channel drivers that support
it
R/O videonativeformat format used natively for video
When I put this in a dialplan with NoOps and called channel macros, I can
kind of get what you're describing:
[from-external-pbxtel]
exten = 491,1,NoOp(${CHANNEL(audioreadformat)})
exten = 491,n,NoOp(${CHANNEL(audiowriteformat)})
exten = 491,n,NoOp(${CHANNEL(audionativeformat)})
exten = 491,n,Dial(SIP/491,20,M(logcodec))
exten = 491,n,Hangup
[macro-logcodec]
exten = s,1,NoOp(${CHANNEL(audioreadformat)})
exten = s,n,NoOp(${CHANNEL(audiowriteformat)})
exten = s,n,NoOp(${CHANNEL(audionativeformat)})
Console output is:
-- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:4] Dial(IAX2/pbxtel-01-5,
SIP/491|20|M(logcodec)) in new stack
-- Called 491
-- SIP/491-0a16d1c0 is ringing
-- SIP/491-0a16d1c0 answered IAX2/pbxtel-01-5
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/491-0a16d1c0, slin) in
new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/491-0a16d1c0, slin) in
new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/491-0a16d1c0, gsm) in new
stack
== Spawn extension (from-external-pbxtel, 491, 4) exited non-zero on
'IAX2/pbxtel-01-5'
-- Hungup 'IAX2/pbxtel-01-5'
This is a call coming in as ulaw over IAX2, then going to a SIP softphone
configured for only gsm.
Hope that helps.
--
j.
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RE: [asterisk-users] SIP Problems continue...
I spoke to soon. Not an hour into the day this morning and we locked up. I'm back to sip debug enable have turned sip history on, get me the bug number and I'll contribute there. Thanks, Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams Sent: Thursday, May 10, 2007 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP Problems continue... Well, I removed and reinstalled Asterisk Zaptel last night. We haven't had one lock up and we've had zero ghost channels kicking around. I copied my config files straight over, so I'm certain it's not a dialplan issue (I was thinking the same thing you were, and I started throwing hangup statements all over the place). My best guess, I had something conflicting with an older version/SVN that was causing grief. We haven't had a day with zero crashes in 2 weeks, and it was progressively getting worse where we were to the point of 4-5 crashes a day. Going an entire day with no crashes is extremely promissing. I do have a lot of data I captured that I could contribute, but I'm not sure we had the exact same problem. ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: Thursday, May 10, 2007 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... 9 maj 2007 kl. 18.14 skrev Ken Williams: SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our old phone system if I can't find a fix in the near future. When the SIP channel locks up the only fix is to restart Asterisk. SIP RELOAD RELOAD CHAN_SIP do no good. Here's a few things I've noticed and changes I've made in hopes of making it better. First, I've currently got 71 active SIP channels when only 2 people are on the phone. This doesn't happen every time, but could be part of the cause. The 'ghost' channels are all INVITES, how do I clear these without rebooting the system? 10.200.26.116716 0a2a959d3d3 00102/0 unkn No Init: INVITE 10.200.26.115715 1dee947d485 00102/0 unkn No Init: INVITE 10.200.26.104704 28808764699 00102/0 unkn No Init: INVITE 10.200.26.104704 36d3e88f59c 00102/0 unkn No Init: INVITE 10.200.26.104704 0e00060800d 00102/0 unkn No Init: INVITE There is an open bug report on this in the bug tracker already. I need your help to find what's causing this issue and provided I can get proper information from you, will spend time locating the bug. First, enable SIP history and catch history for these calls that hang with sip show history Secondly, check the dialplan and tell me more. Where are you calling, why doesn't the other end respond? It's usually calls where we retransmit a number of times and then forget to destroy the calls. If I can get a better description so I can repeat this, I'm sure the bug can be killed. In the bug tracker, there's a patch that will help you. However, until I find more exact information about the nature of these calls, I'm unwilling to commit it. To commit a fix to a poorly defined issue is usually causing more issues, something I can do in trunk but don't want to do in release code. Please send the required information directly to my e-mail address and I'll take a look. Thank you for your assistance with this bug. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM410P
Hello, Here is my config: - /etc/zaptel.conf # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 /etc/asterisk/zapata-channels.conf You need to #include zapata-channels.conf in your zapata.conf ; signalling = pri_cpe is USER ; signalling = pri_net is NETWORK group = 1 switchtype = national signalling = pri_net context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_net context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_net context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 I use FreePBX as my front-end to route calls... so I just assign the trunk groups which I want to use... Regards, Nitesh Alexandre VERNIOL wrote: HI all, Does some one can give me his configuration (zapta.conf, zaptel.conf, sample for extensions.conf) for TDM410P card or similar (1/2/3/4 BRI card) Thanks in advance. Cheers, Alex ___ --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] Need some help with a very simple Queestion..
Ps. Please start new messages from scratch rather then reply to existing ones... (a mistake I've made in the past )-: Woops.. I was ment to remove all that before I posted... :-( Sorry... Best Regards Gavin Spurgeon Systems Administrator Leigh City Technology College [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 -- This message has been scanned for viruses and dangerous content by the Systems @ the LeighCTC, and is believed to be clean. ___ --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] Module wctdm24xxp not found - TDM808P on debian
On 5/11/07, Juliano Fernandes Schroeder [EMAIL PROTECTED] wrote: I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp i get this error FATAL: Module wctdm24xxp not found. FATAL: Error running install command for wctdm24xxp I think i have successfully compiled the zaptel drivers, and the card appears when i do a lspci 02:06.0 Ethernet controller: Digium, Inc. Unknown device 0800 (rev 11) Subsystem: Digium, Inc. Unknown device 0800 Flags: bus master, medium devsel, latency 64, IRQ 3 I/O ports at e800 [size=256] Memory at fe20 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 2000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 I've searched for a solution with no success. Another problem is that when i do a ztcfg -vv i get Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected I don't know if the errors are connected and would appreciate some help. Did you make install in the zaptel source directory? If you didn't, it didn't put the kernel modules in your kernel's module directory and run depmod, so your system doesn't know about the modules. ___ --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 some help with a very simple Queestion..
Hi List, Just a simple question for the list this time.. I need to setup 2 Phones than can Both ring when an incoming call is made to a certain number... I have done this 3,000,000s times with CCM and have no problems with it, But it is the 1st time I have needed to do this with Asterisk. I think it can be done using Queues/Agents but I'm just unsure how do it.. The setup in question is a very small 5 Phones System based on SME 7.1 SAIL (Asterisk web interface) I have a small Sandbox setup here with me to test the test before I need to go set it up on the live system. My test phone is a Grandstream GXP 2000 but I will be using SPA-941's in the Live Environment Any help with this simple question would be great. Best Regards Gavin Spurgeon Systems Administrator Leigh City Technology College [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 - Original Message - From: Alexandre VERNIOL [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 2:43 PM Subject: Re: [asterisk-users] TDM410P Great !!! Thanks a lot !! Nitesh Divecha a écrit : Hello, Here is my config: - /etc/zaptel.conf # T1 Configuration span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,1,0,esf,b8zs bchan=25-47 dchan=48 span=3,1,0,esf,b8zs bchan=49-71 dchan=72 span=4,1,0,esf,b8zs bchan=73-95 dchan=96 /etc/asterisk/zapata-channels.conf You need to #include zapata-channels.conf in your zapata.conf ; signalling = pri_cpe is USER ; signalling = pri_net is NETWORK group = 1 switchtype = national signalling = pri_net context = from-zaptel channel = 1-23 group = 2 switchtype = national signalling = pri_net context = from-zaptel channel = 25-47 group = 3 switchtype = national signalling = pri_net context = from-zaptel channel = 49-71 group = 4 switchtype = national signalling = pri_cpe context = from-zaptel channel = 73-95 I use FreePBX as my front-end to route calls... so I just assign the trunk groups which I want to use... Regards, Nitesh Alexandre VERNIOL wrote: HI all, Does some one can give me his configuration (zapta.conf, zaptel.conf, sample for extensions.conf) for TDM410P card or similar (1/2/3/4 BRI card) Thanks in advance. Cheers, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by the Systems @ the LeighCTC, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Systems @ the LeighCTC, and is believed to be clean. ___ --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 some help with a very simple Queestion..
On Fri, 11 May 2007, Gavin Spurgeon wrote: Hi List, Just a simple question for the list this time.. I need to setup 2 Phones than can Both ring when an incoming call is made to a certain number... exten = 123,1,Dial(SIP/101SIP/102) Gordon Ps. Please start new messages from scratch rather then reply to existing ones... (a mistake I've made in the past )-: ___ --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] Module wctdm24xxp not found - TDM808P on debian
i did, and looking at the output, he processed all the modules into the kernel just fine. On 5/11/07, William Moore [EMAIL PROTECTED] wrote: On 5/11/07, Juliano Fernandes Schroeder [EMAIL PROTECTED] wrote: I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp i get this error FATAL: Module wctdm24xxp not found. FATAL: Error running install command for wctdm24xxp I think i have successfully compiled the zaptel drivers, and the card appears when i do a lspci 02:06.0 Ethernet controller: Digium, Inc. Unknown device 0800 (rev 11) Subsystem: Digium, Inc. Unknown device 0800 Flags: bus master, medium devsel, latency 64, IRQ 3 I/O ports at e800 [size=256] Memory at fe20 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 2000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 I've searched for a solution with no success. Another problem is that when i do a ztcfg -vv i get Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected I don't know if the errors are connected and would appreciate some help. Did you make install in the zaptel source directory? If you didn't, it didn't put the kernel modules in your kernel's module directory and run depmod, so your system doesn't know about the modules. ___ --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 -- Juliano F. Schroeder -- Solucionathica http://www.solucionathica.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
Re: [asterisk-users] Need some help with a very simple Queestion..
Gavin Spurgeon wrote: Ps. Please start new messages from scratch rather then reply to existing ones... (a mistake I've made in the past )-: Woops.. I was ment to remove all that before I posted... Actually, what he's referring to is that posters should start a NEW thread for a new subject. To send a message to the list, click Compose or New or whatever the button is on your particular client (apologies to those using console clients like Mutt) for new messages and enter the list address in the To: field. This means *not* clicking 'reply' to an existing message on the list and then rewriting the subject line (seems like a lot of extra work anyway, doesn't it?) People do this because they can't be bothered to type the list address. That's not hard to solve -- add the address to your address book and create a nickname for it. The reason is that it screws up the message threading. If you are using a threaded reader, or if you are in the archives, you'll have a tree of messages with the original subject line (say, My Asterisk server blew up!) and in the middle of it there'll be something totally unrelated (say, Marmite is good on scones.) Very frustrating if you're trying to read through a thread in the hopes of solving your problem. It's most important for the archives, as these are saved for future reference and are insanely confusing when somebody has piped in to an existing thread. -Stephen- ___ --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] Dealing with 2 SIP providers
Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten = 1234,1,Dial(SIP/providerA) exten = 1234,2,Dial(providerB) exten = 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a total of 60. I want to wait only 30 seconds before the hang up. Also, if ProviderA has a main server and a backup server, am I now forced to have 3 Dial commands, or can I setup ProviderA with host and backuphost in the same SIP entry? Mike ___ --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] Problems with outbound calls through VSP
Bear with me this is a bit long winded. I am having some issues making
automated outbound calls over Broadvoice from my Asterisk 1.4.2 server.
For reference, none of the below issues happen when I make the calls to
VoIP phones attached to the Asterisk server. What I am trying to do is
call, using a .call file, out via the SIP trunk we have setup, and when
the party picks up use AMD to detect if it's reached a human or
machine. If it's human then one message will be played, and if machine
another will be played theoretically after the answering
machine/voicemail is done playing. By the way, I'd like to mention that
this is not at all for spamming, or telemarketing. This is an
appointment reminder service.
from extensions.conf:
[mycontext]
exten = 899,1,Answer
exten = 899,2,Wait(2)
exten = 899,3,AMD
exten = 899,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten = 899,n(mach),WaitForSilence(2500)
exten = 899,n,Playback(were-sorry)
exten = 899,n,Hangup
exten = 899,n(humn),WaitForSilence(500)
exten = 899,n,Playback(welcome)
exten = 899,n,Hangup
The call goes out fine. When I pick it up AMD basically locks up,
although not exactly because as you can see below it does recognize the
HANGUP. However, it will not recognize my voice or dead air no matter
how long I stay on the call to try. If I just let my voicemail pickup
it does the same thing...takes forever for the call to terminate.
Again, this all works as expected when I make the call to a SIP phone
attached to the Asterisk server.
-- Attempting call on SIP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 1)
Channel SIP/sip.broadvoice.com-08bad080 was answered.
-- Executing [EMAIL PROTECTED]:1]
Answer(SIP/sip.broadvoice.com-08bad080, ) in new stack
-- Executing [EMAIL PROTECTED]:2]
AMD(SIP/sip.broadvoice.com-08bad080, ) in new stack
-- AMD: SIP/sip.broadvoice.com-08bad080 (null) (Fmt: 4)
-- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
[800] totalAnalysisTime [5000] minimumWordLength [100]
betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
-- AMD: HANGUP
I did find a solution to this lock up. That was to play a bit of
silence at any point before I actually call AMD (even before Answer works):
[mycontext]
exten = 899,1,Playback(silence/1)
exten = 899,2,Answer
Although I don't particularly like this solution, as I'm just patching
the problem that I still don't understand, plus it adds a little more
delay that confuses the called party.
Also, when I tried this I realized yet another issue, which could be the
underlying cause of the whole thing. No matter what sound it is, no
matter if I use AMD or not, the very first sound that I play results in
a short screech sound before it is played. This happens every time
without fail. If I were to guess, I would say that there is some data
in the audio channel that is not audio data, and is being represented
with that screech sound...but of course that's just a guess.
Any help would be greatly appreciated. Below are some relevant
configuration settings:
sip.conf:
[general]
context=testusers ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
externip=xx.xx.xx.xx
localnet=192.168.1.0/255.255.255.0
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
pedantic=no
register =
[EMAIL PROTECTED]:mysecret:[EMAIL PROTECTED]
[sip.broadvoice.com]
allow=ulaw
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=716XXX
secret=mysecret
username=716XXX
insecure=very
context=from_broadvoice
authname=716XXX
dtmf=inband
dtmfmode=inband
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=yes
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Re: [asterisk-users] Need some help with a very simple Queestion..
I need to setup 2 Phones than can Both ring when an incoming call is made to a certain number... Right.. After a little clicking around and getting a fresh pair of eyes to help me look-over the web interface of SAIL, I found a way of doing what I wanted by adding an Alias and making all inbound calls ring the alias. Thanks For your suggestion Gordon, That also worked by hand editing the .conf files... Best Regards Gavin Spurgeon Systems Administrator Leigh City Technology College [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 -- This message has been scanned for viruses and dangerous content by the Systems @ the LeighCTC, and is believed to be clean. ___ --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] Log CODECS in CDR's
Hi Morgan,
Am Freitag, den 11.05.2007, 10:32 +0100 schrieb Morgan Gilroy:
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729 Dial then negotiates GSM for the outbound
leg,
I want to log both these codecs in a CDR.
At the moment to find the codecs used I have to look though the sip
trace or show channels/show channel (annoying when you have 50+
channels).
Im just trying to find an easier and quicker way to keep track of the
codecs used to help with debug etc.
The closest variable iv found is, ${SIP_CODEC} Set the SIP codec for a
call
Ill see if NoOp (${SIP_CODEC}) shows the codec that was used without me
setting it though I don't think it will.
Iv looked all over and I cant find anything so it looks like I may have
to hack a ast_set_var into app_dial or chan_sip
It is untested, but maybe You can write a little AGI-Script which
accesses some channel vars. Call that AGI as a DeadAGI. A DeadAGI will
be called, if a connection terminates (connect it with the
'h'-Extension, see the wiki). I don't know if the neccessary information
is still alive at this time, but maybe it will do what You want...
HTH,
Karsten
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[asterisk-users] muscionhold error message
hi there guys! how can I eliminate this message? [May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player [May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' This is on debian etch 4.0 asterisk 1.4, it happens quite often everyday and I have to scroll a lot to try to find other error messages. btw can I just put some musica wav files in /var/lib/asterisk/mohmp3 ? that would be great to leave asterisk's processor alone thanks! Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --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 some help with a very simple Queestion..
Stephen Bosch wrote: Gavin Spurgeon wrote: Ps. Please start new messages from scratch rather then reply to existing ones... (a mistake I've made in the past )-: Woops.. I was ment to remove all that before I posted... Actually, what he's referring to is that posters should start a NEW thread for a new subject. To send a message to the list, click Compose or New or whatever the button is on your particular client (apologies to those using console clients like Mutt) for new messages and enter the list address in the To: field. This means *not* clicking 'reply' to an existing message on the list and then rewriting the subject line (seems like a lot of extra work anyway, doesn't it?) People do this because they can't be bothered to type the list address. That's not hard to solve -- add the address to your address book and create a nickname for it. The reason is that it screws up the message threading. If you are using a threaded reader, or if you are in the archives, you'll have a tree of messages with the original subject line (say, My Asterisk server blew up!) and in the middle of it there'll be something totally unrelated (say, Marmite is good on scones.) Is Marmite also available in Ontario, or only Out West? Very frustrating if you're trying to read through a thread in the hopes of solving your problem. It's most important for the archives, as these are saved for future reference and are insanely confusing when somebody has piped in to an existing thread. -Stephen- -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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
Re: [asterisk-users] Strange problem with asterisk
Hi, Vitaly: Vitaly Oborsky wrote: Situation such. There is an asterisk working as office pbx. 6 fxo - 18 fxs ports. All works perfectly, but some times in a week something occurs. Could not catch what exactly yet. But symptoms such. The asterisk infinitely writes the message of a type to broad gullies: WARNING [20757] chan_zap.c: We're Zap/8-1, not ... ZOMBIE. Numbers of channels can change. Because of that that broad gullies get littered fairly promptly, I have not time to see that occured in an instant of the beginning of this event. When the asterisk is in such condition, the appropriating channel does not work, in this case 8. What can it be? asterisk version 1.2.14-BRIstuffed-0.3.0-PRE-1x This looks suspiciously like a Babelfish translation... and I have to admit it's a bit confusing. Can you try rewording it? :\ -Stephen- ___ --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 some help with a very simple Queestion..
You don't need queues to ring two phones, you can simply use the dial command: Dial(SIP/1SIP/10001) -- Would dial SIP extensions 1 and 10001. I would strongly suggest you might want to create a queue for it anyway. One of the things I've noticed in the past (and it's not asterisk's fault - it's a SIP endpoint issue) is that the majority of phones declare themselves busy in their SIP reply whilst they're ringing. So, for example, if two calls come in very close together and both phones are ringing for the first call, the second will receive busy even if the second SIP device could take that call. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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] Dundi and unknown remote peers
Is it possible to allow remote peers to connect to your local DUNDi Asterisk box, even if you don't have them listed in the dundi.conf? I seem to remember something in the sample config file about a [*] entry being possible... One would assume that would cover connections from undefined DUNDi clients. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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] ices low volume
Jon-o Addleman wrote: I'm using the ices command to stream a conference to an icecast server. This is working nicely, for the most part, but the volume is very low. The streamed ogg vorbis audio is much quieter than what I hear in a SIP client, for example (on the same machine with the same audio hardware, of course). Replying to my own question: It appears that's it's not a problem with the ices application. I dumped the audio using a simple EAGI bash script (cat /dev/fd/3 output.raw) and found the same low volume level. I must be misunderstanding something here. Why would the volume be so different through a voip client compared to the audio data dumped from the channel? -- Jon-o Addleman - http://www.redowl.ca ___ --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] Swissvoice IP10s setup
Hi Does anyone have a howto on how to set one of these up on Asterisk or Trix box please? I can make it SIP or MGCP so whatever you have ;-) I have found one page but it isn't really a howto setup Thanks in advance Paul___ --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] Dry Copper Pair
Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over 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
Re: [asterisk-users] Rapid DTMF missing digits
I have actually seen this behaviour on 1.2.x. I always assumed it was just me dialing too fast for the ATA. On 5/11/07, Bryan Laird [EMAIL PROTECTED] wrote: Version 1.4.2 but to be honest I have no reason at all to suspect that this is a problem with the asterisk software. I've able to replicate this from a few different client net connections and a across a few different linksys ata's. Where when you call into the host and enter the extension to connect to you miss the last digit of the extension. Almost every time you miss the last digit of the extension (in a 4 digit extension). My suspicion is simply because of the network we are currently using to host the asterisk box, as a packet dump on the lan segment clearly showed that the ATA transmitted all digits (rfc2833) but the asterisk host only recieved 3 of the 4. The second you dial slower everything works fine; also the lines for voice are clear with no noticeable impairments. I'm more curious if anyone else has ever run into a similar problem and what the resolution was if they found one (IE a sturdier net connection for the asterisk host), or Tweaking the timers on the ata's to slow down how fast and how long they transmit digits. I've done a few different tests and if I use a 'softphone' dialing directly into the server things work perfectly. I can dial as fast as I want, however when I come in through the pstn trunks through the upstream provider I find this problem. has anyone else ever seen this? Or seen a case where mis-matched dtmf modes across multiple providers causes this problem? minor detail on what I referred to as the 'pstn trunks' I have no analog or digital circuts all handoffs are sip. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird Saving Lost Packets since 1994 Have you seen this packet? 101010010 ___ --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] Dry Copper Pair
On Fri, May 11, 2007 at 02:36:46PM -0400, Matt wrote: Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it Yeah; that's called F U pricing. Why would they want to sell you *that*? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
You might be able to try ordering it from a CLEC that can provision it over UNE and sell it for considerably less. Depending on your area, their interconnection agreement, tariffs, etc. So, your mileage may vary. On Fri, 11 May 2007, Matt said something to this effect: Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it around here (Canada) its a tariffed service and I think its about $16 or $18 for the whole thing (both ends). you just have to spend $200 of time on the phone to find someone who knows what you are talking about first. its also referred to as DVACS up here but that's really what's on it, not the pair itself. There may also be some magic used to aggregate the low speed serial channels into a single TDM higher speed circuit. if you are planning on running your own g.hdsl or something like that, I'd love to hear how many cable feet you have and what sort of results you get if you finally get something hooked up. around here you are pretty much limited to adsl, single pair hdsl delivering T1, or analog lines, no more isdn bri's, none of the fancier dsl variations. Other than that you can experiment with dry copper, or try to get fibre if its available. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dry Copper Pair
Let me see. Dry pair, $40 for the circuit. Hardware for each end, $0. Not paying verizon for DSL or PTP T-1 service? Priceless. It's a BANA circuit, btw, in Verizon territory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, May 11, 2007 3:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dry Copper Pair You might be able to try ordering it from a CLEC that can provision it over UNE and sell it for considerably less. Depending on your area, their interconnection agreement, tariffs, etc. So, your mileage may vary. On Fri, 11 May 2007, Matt said something to this effect: Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] A couple of questions for the Mitel gurus(phone-related - not systems)
Nigel, You cannot upgrade a non-dual mode 5220 to SIP. If you are referring to the cable that connects the 5310 to a 5235, that is a standard CAT5 straight-through cable. Barry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel Kendrick Sent: Friday, May 11, 2007 7:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] A couple of questions for the Mitel gurus(phone-related - not systems) Hi Folks, Just in case there are any Mitel gurus here: 1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the SIP firmware? I have inherited one that's Minet only. 2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's lost the connecting lead. Can anyone recommend anywhere in the UK for a replacement lead or confirm the pin-out so I can check whether a generic RJ-RJ lead will work without frying anything. Thanks Nigel Kendrick ___ --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] Dry Copper Pair
Yeah tried that. The CLEC said that one end of the line has to end on their equipment. On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote: You might be able to try ordering it from a CLEC that can provision it over UNE and sell it for considerably less. Depending on your area, their interconnection agreement, tariffs, etc. So, your mileage may vary. On Fri, 11 May 2007, Matt said something to this effect: Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
Who said I wanted to run DSL over it :) On 5/11/07, Smith, Rick [EMAIL PROTECTED] wrote: Let me see. Dry pair, $40 for the circuit. Hardware for each end, $0. Not paying verizon for DSL or PTP T-1 service? Priceless. It's a BANA circuit, btw, in Verizon territory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, May 11, 2007 3:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dry Copper Pair You might be able to try ordering it from a CLEC that can provision it over UNE and sell it for considerably less. Depending on your area, their interconnection agreement, tariffs, etc. So, your mileage may vary. On Fri, 11 May 2007, Matt said something to this effect: Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Dry Copper Pair
How far is the run? I'm wondering what you mean by $0 for hardware? I typically use Ethernet extenders, but it has been a crapshoot on the quality from Verizon. What is a BANA circuit? Finding someone who will even sell it to you has been somewhat of a game as well. From: [EMAIL PROTECTED] on behalf of Smith, Rick Sent: Fri 5/11/2007 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Dry Copper Pair Let me see. Dry pair, $40 for the circuit. Hardware for each end, $0. Not paying verizon for DSL or PTP T-1 service? Priceless. It's a BANA circuit, btw, in Verizon territory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, May 11, 2007 3:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dry Copper Pair You might be able to try ordering it from a CLEC that can provision it over UNE and sell it for considerably less. Depending on your area, their interconnection agreement, tariffs, etc. So, your mileage may vary. On Fri, 11 May 2007, Matt said something to this effect: Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 winmail.dat___ --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] Any other softPBX like Asterisk?
Thank you. I will go through these softwares. Luca Corti [EMAIL PROTECTED] wrote: On Fri, 2007-05-11 at 07:33 -0300, Roberto Pereyra wrote: - OpenPBX - Freeswitch Other: sipX Yet another: Yate http://yate.null.ro/pmwiki/ ciao Luca ___ --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 - Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
Who said I wanted to run DSL over it :) no one - I'm sure you really just want to run 110baud modem over it :) and I'm sure you probably don't want a handful of them between the same 2 locations either. btw - here is an interesting strategy to get fibre or something better than you have at low cost, find out how many analog lines are available on the street in front of you, place an order for N+1 lines. Wait for the installation to happen, then cancel the lines after paying for a month. Depending how saturated the area is, this can be a cheap way to force an upgrade and either get your analog lines delivered on t1 or get fibre to the building etc. On 5/11/07, Smith, Rick [EMAIL PROTECTED] wrote: Let me see. Dry pair, $40 for the circuit. Hardware for each end, $0. Not paying verizon for DSL or PTP T-1 service? Priceless. It's a BANA circuit, btw, in Verizon territory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Friday, May 11, 2007 3:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dry Copper Pair You might be able to try ordering it from a CLEC that can provision it over UNE and sell it for considerably less. Depending on your area, their interconnection agreement, tariffs, etc. So, your mileage may vary. On Fri, 11 May 2007, Matt said something to this effect: Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over it -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ --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] Strange problem with asterisk
On Fri, May 11, 2007 at 05:32:33PM +0300, Vitaly Oborsky wrote: Situation such. There is an asterisk working as office pbx. 6 fxo - 18 fxs ports. All works perfectly, but some times in a week something occurs. Could not catch what exactly yet. But symptoms such. The asterisk infinitely writes the message of a type to broad gullies: WARNING [20757] chan_zap.c: We're Zap/8-1, not ... ZOMBIE. Numbers of channels can change. Because of that that broad gullies get littered fairly promptly, I have not time to see that occured in an instant of the beginning of this event. When the asterisk is in such condition, the appropriating channel does not work, in this case 8. What can it be? asterisk version 1.2.14-BRIstuffed-0.3.0-PRE-1x Could you try a later verssion of bristuff? I seem to recall a bug report for chan_zap (or is it Zaptel) with exactly those symptoms. I cannot find it right now. Anybody? -- 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] Dealing with 2 SIP providers
From: Mike [EMAIL PROTECTED] Date: Fri, 11 May 2007 11:06:35 -0400 Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten = 1234,1,Dial(SIP/providerA) exten = 1234,2,Dial(providerB) exten = 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a total of 60. I want to wait only 30 seconds before the hang up. Like put 15 seconds on each? It's quite hard to understand what exactly the requirements are. Yuan Liu Also, if ProviderA has a main server and a backup server, am I now forced to have 3 Dial commands, or can I setup ProviderA with host and backuphost in the same SIP entry? Mike ___ --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 SIP domain (in LAN or DMZ)?
Hi, In my opinion, you should keep your Asterisk, probably with PSTN Cards, inside your network and just setup an OpenSer or even simpler another Asterisk server on your DMZ. This way you will enable ENUM and URIs for your Clients, and will prevent much better any DoS, intrusion or any other backhole that would let external users places PSTN calls through your server. At the sametime if something goes wrong on outside world, your Lan VoIP going will be kept 99,99% fully functional and let you make and receive calls through PSTN. Good Luck, Marco Mouta Ps. Qualquer coisa apita:) On 5/10/07, Joao Pereira [EMAIL PROTECTED] wrote: Hello I want to use Asterisk to implement a SIP Domain allowing my clients to do URI dialing and receive calls from the Internet through URIs and ENUM. My question is, should I put my Asterisk outside the firewall (in the DMZ) to allow connections to the Internet? Or should I have it inside my local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks Regards Joao Pereira ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dry Copper Pair
Matt et al, Can you still do homebrew PTP T1 in the U.S. this way? I thought this was nixed by the ILEC/CLECs years ago. John Treble Ottawa, Canada From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: May 11, 2007 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dry Copper Pair Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over 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
RE: [asterisk-users] Dry Copper Pair
On Fri, 11 May 2007, John Treble said something to this effect: Can you still do homebrew PTP T1 in the U.S. this way? I thought this was nixed by the ILEC/CLECs years ago. It's logically possible. But if you're trying to do T1 over a single pair, you'd have to break it out using HDSL/PairGain sort of line equipment, since you obviously can't install field repeaters or do any span conditioning yourself. From then on it's a crapshoot and really just depends on whether the copper is of quality, distance, specifications, etc. that can support the specification. There's no way for them to nix that, really, other than possibly keeping load coils or other constraining stuff on the facilities that tends to need to be removed for various high-speed data line / private line applications. -- Alex -- Alex Balashov [EMAIL PROTECTED]___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk crashes
Hello, I have very annoying problem with asterisk 1.4.4: Every evening when I have peak load asterisk crashes, peak load is only over 20-30 sip-to-h323 simultaneous calls. Nothing special in logs after crash. Load average never was higher than 0.3, asterisk never uses more than 12% CPU (according to top). Tried SVN versions - same result. Both h323 and sip peers has only one codec allowed - g729 - so no conversion. There is no conferences, call recordings or something like this - very simple setup. Software config: Linux Slackware 11.0 Kernel 2.6.21.1 Asterisk 1.4.4 (native h323 channel from asterisk tarball) Libpri-1.4.0 Zaptel-1.4.2.1 (using ztdummy for internal sync, no zaptel hardware) pwlib_v1_10_0 openh323_v1_18_0 Hardware config: Intel SE7210TP1 motherboard P4 3GHz HT 1Mb cache CPU 1Gb RAM (dual channel, two same DIMMs from intel recommended list) 80Gb SATA HDD No zaptel hardware or even any PCI cards There isn't overheating and voltage problems with a hardware (controlling over IPMI), this hardware (with another HDD and software versions) worked fine about year with asterisk restarts manually only for a version upgrade. Could somebody point me the way to debug this problem? Thank you! -- Sincerely, Elman Efendiyev ___ --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] Problems with outbound calls through VSP
Update:
I was able to obtain another VSP to try and rule out Broadvoice. Seems
that either my Broadvoice settings, or something on their end is causing
the brief screech noise upon playing the first sound.
However, with this new VSP I still have the AMD (Answering Machine
Detect) problem where it locks up unless I play some sound before
calling AMD. So my modified question is, has anyone ever had a problem
with AMD through a VSP (SIP, in this case). And it does *not* lock up
when calling phones local to the server.
Christopher Robinson wrote:
Bear with me this is a bit long winded. I am having some issues
making automated outbound calls over Broadvoice from my Asterisk 1.4.2
server. For reference, none of the below issues happen when I make
the calls to VoIP phones attached to the Asterisk server. What I am
trying to do is call, using a .call file, out via the SIP trunk we
have setup, and when the party picks up use AMD to detect if it's
reached a human or machine. If it's human then one message will be
played, and if machine another will be played theoretically after the
answering machine/voicemail is done playing. By the way, I'd like to
mention that this is not at all for spamming, or telemarketing. This
is an appointment reminder service.
from extensions.conf:
[mycontext]
exten = 899,1,Answer
exten = 899,2,Wait(2)
exten = 899,3,AMD
exten = 899,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten = 899,n(mach),WaitForSilence(2500)
exten = 899,n,Playback(were-sorry)
exten = 899,n,Hangup
exten = 899,n(humn),WaitForSilence(500)
exten = 899,n,Playback(welcome)
exten = 899,n,Hangup
The call goes out fine. When I pick it up AMD basically locks up,
although not exactly because as you can see below it does recognize
the HANGUP. However, it will not recognize my voice or dead air no
matter how long I stay on the call to try. If I just let my voicemail
pickup it does the same thing...takes forever for the call to
terminate. Again, this all works as expected when I make the call to
a SIP phone attached to the Asterisk server.
-- Attempting call on SIP/[EMAIL PROTECTED] for
[EMAIL PROTECTED]:1 (Retry 1)
Channel SIP/sip.broadvoice.com-08bad080 was answered.
-- Executing [EMAIL PROTECTED]:1]
Answer(SIP/sip.broadvoice.com-08bad080, ) in new stack
-- Executing [EMAIL PROTECTED]:2]
AMD(SIP/sip.broadvoice.com-08bad080, ) in new stack
-- AMD: SIP/sip.broadvoice.com-08bad080 (null) (Fmt: 4)
-- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence
[800] totalAnalysisTime [5000] minimumWordLength [100]
betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
-- AMD: HANGUP
I did find a solution to this lock up. That was to play a bit of
silence at any point before I actually call AMD (even before Answer
works):
[mycontext]
exten = 899,1,Playback(silence/1)
exten = 899,2,Answer
Although I don't particularly like this solution, as I'm just patching
the problem that I still don't understand, plus it adds a little more
delay that confuses the called party.
Also, when I tried this I realized yet another issue, which could be
the underlying cause of the whole thing. No matter what sound it is,
no matter if I use AMD or not, the very first sound that I play
results in a short screech sound before it is played. This happens
every time without fail. If I were to guess, I would say that there
is some data in the audio channel that is not audio data, and is being
represented with that screech sound...but of course that's just a guess.
Any help would be greatly appreciated. Below are some relevant
configuration settings:
sip.conf:
[general]
context=testusers ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
externip=xx.xx.xx.xx
localnet=192.168.1.0/255.255.255.0
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
pedantic=no
register =
[EMAIL PROTECTED]:mysecret:[EMAIL PROTECTED]
[sip.broadvoice.com]
allow=ulaw
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=716XXX
secret=mysecret
username=716XXX
insecure=very
context=from_broadvoice
authname=716XXX
dtmf=inband
dtmfmode=inband
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=yes
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RE: [asterisk-users] Dry Copper Pair
On Fri, 11 May 2007, John Treble said something to this effect: Can you still do homebrew PTP T1 in the U.S. this way? I thought this was nixed by the ILEC/CLECs years ago. It's logically possible. But if you're trying to do T1 over a single pair, you'd have to break it out using HDSL/PairGain sort of line equipment, since you obviously can't install field repeaters or do any span conditioning yourself. From what I know about it - even with the hdsl you are only going to get it to work at full speed over about 1 cable ft, then you need a repeater of some sort - if it was just a raw T1, you're not going to get anywhere near the 1 ft to start with. the dry copper is a cheap install since they DON'T do the line conditioning - remove load coils, etc, but if your reach was only 1ft to start with you're not likely to have load coils etc anyway, and if the line is that bad where its got grounds or shorts you would be within your rights to demand that be fixed even for dry copper. The question is really can you get dry copper short enough cable ft to span the locations you need and still work with whatever hardware you want to throw on the ends of it ? as far as the conditioning, you could probably even get 2ft without coils if the CO is halfway in the middle since the coils would be based on the radius from the CO in the first place, but then again is your hardware going to reach that distance and be able to maintain any sort of decent transfer rate ? again, I'm interested to know anyone whose actually done this, and what the results were, since I have been thinking of the same thing for a while. From then on it's a crapshoot and really just depends on whether the copper is of quality, distance, specifications, etc. that can support the specification. There's no way for them to nix that, really, other than possibly keeping load coils or other constraining stuff on the facilities that tends to need to be removed for various high-speed data line / private line applications. -- Alex -- Alex Balashov [EMAIL PROTECTED]___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
So we know, and I know, that a dry copper pair has no load coils, etc. Generally sells for about $20/line.. sometimes less. Is there something that iLEC will sell that has load coils in it? Like say, if I wanted to run voice over it, and didn't care about data? IE.. I know this is VoIP, but say I wanted to put an analog extension someplace.Is there a cheap alternative I could hook between me and the remote location, going analog all the way? On 5/11/07, Jon Pounder [EMAIL PROTECTED] wrote: On Fri, 11 May 2007, John Treble said something to this effect: Can you still do homebrew PTP T1 in the U.S. this way? I thought this was nixed by the ILEC/CLECs years ago. It's logically possible. But if you're trying to do T1 over a single pair, you'd have to break it out using HDSL/PairGain sort of line equipment, since you obviously can't install field repeaters or do any span conditioning yourself. From what I know about it - even with the hdsl you are only going to get it to work at full speed over about 1 cable ft, then you need a repeater of some sort - if it was just a raw T1, you're not going to get anywhere near the 1 ft to start with. the dry copper is a cheap install since they DON'T do the line conditioning - remove load coils, etc, but if your reach was only 1ft to start with you're not likely to have load coils etc anyway, and if the line is that bad where its got grounds or shorts you would be within your rights to demand that be fixed even for dry copper. The question is really can you get dry copper short enough cable ft to span the locations you need and still work with whatever hardware you want to throw on the ends of it ? as far as the conditioning, you could probably even get 2ft without coils if the CO is halfway in the middle since the coils would be based on the radius from the CO in the first place, but then again is your hardware going to reach that distance and be able to maintain any sort of decent transfer rate ? again, I'm interested to know anyone whose actually done this, and what the results were, since I have been thinking of the same thing for a while. From then on it's a crapshoot and really just depends on whether the copper is of quality, distance, specifications, etc. that can support the specification. There's no way for them to nix that, really, other than possibly keeping load coils or other constraining stuff on the facilities that tends to need to be removed for various high-speed data line / private line applications. -- Alex -- Alex Balashov [EMAIL PROTECTED]___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ --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] Dry Copper Pair
On Fri, 11 May 2007, Matt said something to this effect: Is there something that iLEC will sell that has load coils in it? Like say, if I wanted to run voice over it, and didn't care about data? I don't know that they'd necessarily sell you anything with load coils *per se*, especially since the general trend is to remove them as xDSL becomes more pervasive, etc. But if you just wanted to run an analog line, there's no reason why you couldn't just put an FXO adaptor on one end and an analog phone on the other in theory. As has been duly noted, actual practice may vary depending on the nature of the analog line, whose physical build you have no real control over. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
On 5/11/07, John Treble [EMAIL PROTECTED] wrote: Matt et al, Can you still do homebrew PTP T1 in the U.S. this way? I thought this was nixed by the ILEC/CLECs years ago. Not according to Verizon (in my area anyhow), We tried it and it didn't work. The verizon technician insisted it wasn't real PTP copper and therefore anything but analog voice might/should not work. John Treble Ottawa, Canada From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: May 11, 2007 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dry Copper Pair Hi, Does anyone know of a way to get a dry copper pair (also known as an alarm line) from Verizon for less than $20/end? I know we have been able to get them, but they come out to $40/month for a circuit.. and there's no dial-tone over 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
On Fri, 11 May 2007, C F said something to this effect: Not according to Verizon (in my area anyhow), We tried it and it didn't work. The verizon technician insisted it wasn't real PTP copper and therefore anything but analog voice might/should not work. What is PTP copper? Unless it's an issue of gauge. But as far as I know, it's not. All the standard copper used for POTS can be used for a T1 from a physical point of view, other aspects of conditioning/load coils/etc/etc not withstanding. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, 11 May 2007, C F said something to this effect: Not according to Verizon (in my area anyhow), We tried it and it didn't work. The verizon technician insisted it wasn't real PTP copper and therefore anything but analog voice might/should not work. What is PTP copper? Unless it's an issue of gauge. But as far as I know, it's not. All the standard copper used for POTS can be used for a T1 from a physical point of view, other aspects of conditioning/load coils/etc/etc not withstanding. You are right, but that was not what I meant, in order for one to be able to provision their own T1 over a pair of copper, the line has to allow all traffic over all frequencies pass thru it. Which these lines do not, since they are simply not just one long copper pair simply cross connected. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, 11 May 2007, C F said something to this effect: Not according to Verizon (in my area anyhow), We tried it and it didn't work. The verizon technician insisted it wasn't real PTP copper and therefore anything but analog voice might/should not work. What is PTP copper? Unless it's an issue of gauge. But as far as I know, it's not. All the standard copper used for POTS can be used for a T1 from a physical point of view, other aspects of conditioning/load coils/etc/etc not withstanding. You are right, but that was not what I meant, in order for one to be able to provision their own T1 over a pair of copper, the line has to allow all traffic over all frequencies pass thru it. Which these lines do not, since they are simply not just one long copper pair simply cross connected. that's what dry copper is supposed to be, just a cross connect between 2 pairs out of the CO. ie not even battery, line test equipment, or anything else hanging off it at the CO. any restriction should be purely a function of the inductance/capacitance of the wire and the connections and nothing else - anything else and you didn't get dry copper in the first place. just out of curiousity - anyone ever hijack pairs and get away with it ? (do your own cross connects on the street and utilize some crossconnect all within one branch of F1 cable out of the CO ?) I've been tempted in the past, and know that at least around here I would probably get away with it for quite some time before anyone actually cared enough to investigate. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
On Fri, 2007-05-11 at 18:44 -0400, Jon Pounder wrote: On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, 11 May 2007, C F said something to this effect: Not according to Verizon (in my area anyhow), We tried it and it didn't work. The verizon technician insisted it wasn't real PTP copper and therefore anything but analog voice might/should not work. What is PTP copper? Unless it's an issue of gauge. But as far as I know, it's not. All the standard copper used for POTS can be used for a T1 from a physical point of view, other aspects of conditioning/load coils/etc/etc not withstanding. You are right, but that was not what I meant, in order for one to be able to provision their own T1 over a pair of copper, the line has to allow all traffic over all frequencies pass thru it. Which these lines do not, since they are simply not just one long copper pair simply cross connected. that's what dry copper is supposed to be, just a cross connect between 2 pairs out of the CO. ie not even battery, line test equipment, or anything else hanging off it at the CO. any restriction should be purely a function of the inductance/capacitance of the wire and the connections and nothing else - anything else and you didn't get dry copper in the first place. just out of curiousity - anyone ever hijack pairs and get away with it ? (do your own cross connects on the street and utilize some crossconnect all within one branch of F1 cable out of the CO ?) I've been tempted in the past, and know that at least around here I would probably get away with it for quite some time before anyone actually cared enough to investigate. Hmmm, I can see cross connecting an F1 to the F2 to your home/business, but you would have to have a friend @ the CO to make anything of use on it right? Someone has to connect it to their frame in the CO, or xconnect it to another F1 out?? If there is a telco with live dialtone on F1 unprovisioned pairs, I would be shocked (or want to move there :) ) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
On Friday 11 May 2007 5:45 pm, Jon Pounder wrote: again, I'm interested to know anyone whose actually done this, and what the results were, since I have been thinking of the same thing for a while. I'd run about two dozen of these things using a variety of equipment. Pairgain SDSL modems (300S), Flowpoint 2200s, Speedstream something-or-others... hell we even used the flowpoints and speedstreams with an SDSL DSLAM. It works reasonably well in-town, and gets you around a megabit to two, depending on distance. lowest speed I did was about 384kbps, and highest was 2048. All these rates are symmetrical, BTW. In Canada you ask for either a Class A signal channel or a dry pair, depending on whether you are talking to the voice or data guys. You need to get in good with the local tech, too, because if there *ARE* coils, Bell will NOT remove them for you, on the record. The voice circuits have an identifier starting with TVCSNA, and the data circuits CCLADA. These days though, we just order nekkid DSL and get dialtone but no ability to dial anything but 911, and the line's connected to their DSLAM. -A. ___ --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] Dealing with 2 SIP providers
Yeah ok. That doesn't help. What I mean is I want a call to go out on ProviderA, UNLESS it's down and then go to ProviderB. I want it to ring 30 seconds and then Hangup if nobody has answers. I DON'T want to dial both, only one or the other. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, May 11, 2007 17:03 To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Dealing with 2 SIP providers From: Mike [EMAIL PROTECTED] Date: Fri, 11 May 2007 11:06:35 -0400 Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten = 1234,1,Dial(SIP/providerA) exten = 1234,2,Dial(providerB) exten = 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a total of 60. I want to wait only 30 seconds before the hang up. Like put 15 seconds on each? It's quite hard to understand what exactly the requirements are. Yuan Liu Also, if ProviderA has a main server and a backup server, am I now forced to have 3 Dial commands, or can I setup ProviderA with host and backuphost in the same SIP entry? Mike ___ --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] Dry Copper Pair
Quoting Greg Oliver [EMAIL PROTECTED]: On Fri, 2007-05-11 at 18:44 -0400, Jon Pounder wrote: On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Fri, 11 May 2007, C F said something to this effect: Not according to Verizon (in my area anyhow), We tried it and it didn't work. The verizon technician insisted it wasn't real PTP copper and therefore anything but analog voice might/should not work. What is PTP copper? Unless it's an issue of gauge. But as far as I know, it's not. All the standard copper used for POTS can be used for a T1 from a physical point of view, other aspects of conditioning/load coils/etc/etc not withstanding. You are right, but that was not what I meant, in order for one to be able to provision their own T1 over a pair of copper, the line has to allow all traffic over all frequencies pass thru it. Which these lines do not, since they are simply not just one long copper pair simply cross connected. that's what dry copper is supposed to be, just a cross connect between 2 pairs out of the CO. ie not even battery, line test equipment, or anything else hanging off it at the CO. any restriction should be purely a function of the inductance/capacitance of the wire and the connections and nothing else - anything else and you didn't get dry copper in the first place. just out of curiousity - anyone ever hijack pairs and get away with it ? (do your own cross connects on the street and utilize some crossconnect all within one branch of F1 cable out of the CO ?) I've been tempted in the past, and know that at least around here I would probably get away with it for quite some time before anyone actually cared enough to investigate. Hmmm, I can see cross connecting an F1 to the F2 to your home/business, but you would have to have a friend @ the CO to make anything of use on it right? Someone has to connect it to their frame in the CO, or xconnect it to another F1 out?? If there is a telco with live dialtone on F1 unprovisioned pairs, I would be shocked (or want to move there :) ) well actually there is dialtone on the unprovisioned pairs for the most part, but you can only dial repair, the telco office or 911 on them. I am not sure if its all pairs or just pairs that had a line provisioned at one time. ANAC just replys with some error message if you try to determine the phone number of the line. What I am talking about though is if you want to run dsl or some other highspeed type of thing or just an analog pair to a neighbour, or another office in the same neighbourhood, complex etc. All you do is put your tone generator on an empty pair at both locations trace down till you find them in the same F1/F2 box, and jump across them. (no connection to or through the CO, but only possible if both areas are served by the same F1 cable.) Around here at least, the worker who actually gets the work order for an analog install is told the frame port and corresponding F1 pair, and they just find a free F2 pair and use it, so unless they happened to notice the cross connect between 2 F2 pairs, or even noticed it and cared, who would know ? Actually it would probably take some investigation to even tell if its a legitimate bridge tap or the left overs of one or just something that is not supposed to be there at all. In a world of if its not broke don't touch it, it would likely never get touched. Even on a lower level, if you want cable between immediate neighbours, just make a cross connect at the nearest pedestal or overhead box if you both are served from it and have a spare pair in your lateral cables. Here's some food for thought - around here at least where there is buried telco fibre, the splices are done in pedestals that don't even have locks on the doors, just a screen door type latch, might keep a racoon out but that would even be pushing it. The copper is a little more secure, you have to carry a nutdriver to give the latch a quarter turn. I guess if you are resourceful enough to have a nutdriver, they trust you poking around in their boxes. Wear a hardhat and toolbelt with a butt set hanging off it, and you'll easily penetrate the collective :) I've had many a conversation with a telco installer and for the most part if you know what you're talking about they practically invite you to help yourself if you want to poke around, modify your cabling etc., just don't say they told you that or complain if you break 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/
Re: [asterisk-users] Dry Copper Pair
On Friday 11 May 2007 7:46 pm, Jon Pounder wrote: well actually there is dialtone on the unprovisioned pairs for the most part, but you can only dial repair, the telco office or 911 on them. I am not sure if its all pairs or just pairs that had a line provisioned at one time. ANAC just replys with some error message if you try to determine the phone number of the line. If you're talking about for DSL use (i.e. connecting to a BAS and using resold DSL service) then yeah, there's almost always dialtone and you can only call the numbers listed. Dry copper (two pairs cross-connected at the CO) has nothing on it. No battery, nothing. Loading coils will be present if one of the loops is exceptionally long, but otherwise it's just as if you'd run the copper between the locations yourself. Where I was located (Listowel, Ontario) we seemed to get better speed vs distance compared to the equipment's ratings, but we chalked that up to having heavier gauge wire in the copper plant (small rural town) and thus less losses in the lines, not to mention possibly a lot fewer competing signals in the trunks. What I am talking about though is if you want to run dsl or some other highspeed type of thing or just an analog pair to a neighbour, or another office in the same neighbourhood, complex etc. All you do is put your tone generator on an empty pair at both locations trace down till you find them in the same F1/F2 box, and jump across them. (no connection to or through the CO, but only possible if both areas are served by the same F1 cable.) Around here at least, the worker who Bell was HIGHLY adverse to this, as it played havoc with their planning, at least according to them. We were only able to have them cross-connected at a pedestal in one (early on) loop; all others were REQUIRED to run through the CO, which often added too much distance for us to make it useful. F2 pairs, or even noticed it and cared, who would know ? Actually it would probably take some investigation to even tell if its a legitimate bridge tap or the left overs of one or just something that is not supposed to be there at all. In a world of if its not broke don't touch it, it would likely never get touched. That's the other thing we ran into from time to time: bridge taps. Loops that should have gotten an easy 1.5meg wouldn't sync at all, and eventually the culprit was found to be a 5km tap run off to some new subdivision. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
On Fri, May 11, 2007 at 07:46:18PM -0400, Jon Pounder wrote: Wear a hardhat and toolbelt with a butt set hanging off it, and you'll easily penetrate the collective :) I've had many a conversation with a telco installer and for the most part if you know what you're talking about they practically invite you to help yourself if you want to poke around, modify your cabling etc., just don't say they told you that or complain if you break it It's really the can-wrench; lots of people have butt-sets these days. A real E-4 doesn't hurt either. And no one wears a hat unless they're on a new-construction site or up a ladder... Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --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] Dealing with 2 SIP providers
What I mean is I want a call to go out on ProviderA, UNLESS it's down and
then go to ProviderB.
I want it to ring 30 seconds and then Hangup if nobody has answers.
This one's actually a bit more complicated than it first seems, since you need
to know how each provider reports status when it's unavailable. We run the
following AEL macros to achieve something similar:
(apologies to the list for the big chunk of code below - I'm not sure how
well/if the list handles attachments)
// DIAL NUMBER (with a range of routing options)
macro outbound (number, route1, route2, route3, route4, route5) {
// set correct outbound caller id
if (${LEN(${CALLERID(number)})} 10 ${LEN(${CALLERID(number)})} 0)
{
if (${LEN(${DB(callerid/${CDR(accountcode)})})} 9) {
CALLERID(number)=${DB(callerid/${CDR(accountcode)})};
} else
Set(CALLERID(number)=);
};
dialstart:
switch (${route1}) {
case dundi:
if (${number:0:2} = 00) {
dundi-e164 (${number:2});
} else if (${number:0:1} = 0) {
dundi-e164 (44${number:1});
} else
dundi-e164 (${number});
break;
case provider1:
dialout (IAX2/provider1/${number});
break;
case provider2:
dialout (IAX2/provider2/${number});
break;
case provider3:
dialout (IAX2/provider3/${number});
break;
case pstn:
dialout (Zap/g1/${number});
break;
default:
NoOp (invalid route: ${route1});
};
if (${LEN(${route2})} 0) {
route1=${route2};
} else {
Playtones (congestion);
Congestion ();
};
if (${LEN(${route3})} 0)
route2=${route3};
if (${LEN(${route4})} 0)
route3=${route4};
if (${LEN(${route5})} 0)
route4=${route5};
goto dialstart;
};
// DIAL NUMBER (ignoring anything except busy)
macro dialout (dialstring) {
Dial (${dialstring},,TW);
switch (${DIALSTATUS}) {
case BUSY:
Playtones (busy);
Busy ();
break;
case CONGESTION:
Playtones (busy);
Busy ();
break;
};
};
You can then dial from your main dialplan something like this for UK
landlines:
exten = _0[12]X,1,Macro(outbound,${EXTEN},provider1,provider2,pstn)
The dialout macro ignores any responses from the SIP/IAX provider except Busy
or Congestion (we have a provider which provides congestion when the dialled
number is busy, that's why it's there). So, if the provider's server is
unavailable (through qualify=yes or whatever), it'll fall through as channel
status unknown and loop onto the next provider.
On an outbound call made from one of your users, why would you want a 30 second
timeout? Surely you'd want to keep ringing the callee until the caller (i.e.
your user) loses interest and hangs up their device? The length of time for a
device to be rung before doing something else is usually determined by the
recipient, not the initiator.
Hope that helps.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons
___
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Re: [asterisk-users] Dealing with 2 SIP providers
Mike wrote: Yeah ok. That doesn't help. What I mean is I want a call to go out on ProviderA, UNLESS it's down and then go to ProviderB. I want it to ring 30 seconds and then Hangup if nobody has answers. I DON'T want to dial both, only one or the other. Mike Mike, You had it correct in your original post. exten=s,1,Dial(Sip/111|30|m) ; === Try this one. If it answers, exten=s,2,Dial(Sip/112|30|m) you don't go to s,2. -- Warm Regards, Lee ___ --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] muscionhold error message
pedro noticioso wrote: hi there guys! how can I eliminate this message? [May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506 monmp3thread: Unable to spawn mp3player [May 11 11:09:06] WARNING[7039]: res_musiconhold.c:424 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' I'm no expert, but haven't seen other replies here, so I'll throw out my suggestions. I don't have that package (still using testing's 1.2 package) but I expect the /etc/asterisk/musiconhold.conf uses mp123 to process the music on hold files. That's not Free software though (and isn't really maintained anymore, I think...), so I don't believe it's included in Debian, though you can get it from http://www.debian-multimedia.org/ if not. You could also just change the musiconhold.conf file I expect. This is on debian etch 4.0 asterisk 1.4, it happens quite often everyday and I have to scroll a lot to try to find other error messages. btw can I just put some musica wav files in /var/lib/asterisk/mohmp3 ? that would be great to leave asterisk's processor alone You can use pretty much anything, since you can define a decoder in the musiconhold.conf file. http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf has some more info, though it's pretty confusing since it has info for different versions of asterisk, and some discussion interspersed... -- Jon-o Addleman - http://www.redowl.ca ___ --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] muscionhold error message
I forgot to add that the built-in support for playing mp3s which replaced, for some people, the mp123 program, requires asterisk-addons, which also isn't packaged for debian! There are other possibilities though. I think you could use mp321 plus sox to convert to the proper sound format, for example. -- Jon-o Addleman - http://www.redowl.ca ___ --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] RTP Mixer
Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. ___ --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 some help with a very simple Queestion..
Drew Gibson wrote: Stephen Bosch wrote: Gavin Spurgeon wrote: Ps. Please start new messages from scratch rather then reply to existing ones... (a mistake I've made in the past )-: Woops.. I was ment to remove all that before I posted... Actually, what he's referring to is that posters should start a NEW thread for a new subject. To send a message to the list, click Compose or New or whatever the button is on your particular client (apologies to those using console clients like Mutt) for new messages and enter the list address in the To: field. This means *not* clicking 'reply' to an existing message on the list and then rewriting the subject line (seems like a lot of extra work anyway, doesn't it?) People do this because they can't be bothered to type the list address. That's not hard to solve -- add the address to your address book and create a nickname for it. The reason is that it screws up the message threading. If you are using a threaded reader, or if you are in the archives, you'll have a tree of messages with the original subject line (say, My Asterisk server blew up!) and in the middle of it there'll be something totally unrelated (say, Marmite is good on scones.) Is Marmite also available in Ontario, or only Out West? As far as I know, Marmite is available all across this land, from sea to sea to sea. Three cheers for Marmite. -Stephen- ___ --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
