Re: [Asterisk-Users] Call Center with No TDM components
On Wed, 19 Apr 2006, Abhimanyu Rapria wrote: Transcoding and Recording is being done at VICIDIAL/ASTERISK Dialer and load average is 1.5 for 12 agents and pacing of 1.1 to 1.2 What is the average CPU utilization you observe with these load averages? Regards, Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
Hi Paul, Thanks for the message! On Sun, 16 Apr 2006, Paul Hewlett wrote: [...] I am curious.. Have you tried disabling CPU1 by setting isolcpus=1 on the kernel command line ? This will make the kernel ignore the second CPU - you can then run asterisk on it by using the taskset command (from schedutils) taskset 0x0001 asterisk -p and asterisk wlll run on a CPU all on its own. I was about to try this and wondered if you might give it a try and report back. I haven't done this yet. Once we have physical access to the machine, I'll make sure we try this out and see what difference it makes. Cheers! Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
On Mon, 17 Apr 2006, stoffell wrote: Interesting. Now 'why' do they suggest it, is it because older IO-APIC are 'broken' on some boards? I'm very curious as to 'why', [...] Most likely this is why. Regards, Gerald ___ --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] Digium cards, so disappointing !
Hi Steve, Thank you for your very enlightening message! On Sat, 15 Apr 2006, Steve Underwood wrote: [...] modem it must be applied end to end by the modems themselves. The real killer, though, is imperfect timing. [...] and its not always always available within a PC. PCs are designed around best efforts handling of data. They don't handle continuous streaming of media well, even if the data rate is fairly low. They handle it especially badly if latency must be kept low, as is the case with I have come to understand and appreciate this fact more and more through painful experience. [...] That said, a well design PC environment can achieve the timing needed for FAX calls, as long as you don't load it up too much. In your opinion, short of re-engineering the PC, is there anything that can be done to step up the timing accuracy (and hence up the real-time performance) of the PC? What [hardware-based] technical action would you think can up the real-time performance of the PC? Regards, Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
On Tue, 11 Apr 2006, Andrew Kohlsmith wrote: Please do not open your mouth to spout nonsense if you do not know what you're talking about. [...] Again, if the IO-APIC is reporting that the card is on its own IRQ, it really, truly, honestly *IS* on its own IRQ. The reason that it is suggested to disable the IO-APIC is that on many low-end systems, the IO-APIC is plain old broken and causing other issues. I don't think I've run across a system board in the last year or two with that issue, though. It's always been on older P3 and early P4 systems. Allow me to comment that Digium actually recommends turning off APIC and using lspci -vb to troubleshoot this kind of shared-interrupt problem. Cheers, Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] te110p and interrupts
Hi, I've been battling with a similar issue: a) I wrote a script to periodically run the command cat /proc/interrupts and figure out the interrupts per second. I run this script for over 24 hours and never once did the difference between the preceeding and succeeding interrupt counts go below 1005 (wierd result because of (b) below); b) zttest was reporting very bad results; c) lspci -vb was reporting that the TE110P shared an IRQ with the Gigabit Ethernet Card (IRQ 11) d) lspci -vv was reporting that the TE110P was on an IRQ of its own (IRQ 24) probably because of APIC (wierd because of (c) above); e) Users reported intermittent bad audio; Below are the [experimental] steps I took: a) I'm running a Dual 3.2 GHz machine - the network card is services by CPU0 - I set the smp_affinity value for the Digium card to be CPU1 b) I disabled the userland 'irqbalance' process which keeps switching the Digium card between the CPUs c) I increased the PCI LATENCY_TIMER value for the TE110P to a value higher than the Gigabit Card. So far, things are looking quite good - zttest is reporting very encouraging worst-case figures when run over a period of over an hour (it reports 99.98% worst case at off peak time and 99.77% when run during the busy hour). Ultimately when I have physical access to the machine, I will change the PCI slots to see if getting lspci -vb to report that the card is on its own IRQ will improve performance further. Cheers, Gerald. ___ --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 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Remco Barende wrote: Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Have you by chance set immediate to yes? IIRC, there's a feature that will send you to the configured context as soon as you pick up your phone (this is in zapata.conf). Might be worth checking that out. Cheers, Gerald. ___ --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 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Remco Barende wrote: I have but only for the phone line, it is immediately after: signalling=fxs_ks immediate=yes What I actually meant is that you should turn this off if you don't need the functionality. Most likely you are defining the extension channel after the phone line thus it is inheriting the setting as well. Gerald. ___ --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] New Application: Broadcast
Hi Steve, On Thu, 13 Oct 2005, Steve Daniels wrote: What excatly does it do? What messages does it send out? And what software needs to be configured to listen for these messages? Bret explained mostly what the software does in a basic use case where you would like a nice window to pop up with say the caller id details of an incoming call. With this same software, you may selectively broadcast messages for example, you may only want the sales crew to see information about a given caller and not other groups. For example: [sales-context] exten = s,1,Answer exten = s,2,Broadcast(This is a sales call|group=sales) exten = s,3,Dial(whatever) In such a case, you will need to have configured the sales computers with a group attribute set to sales for example: [192.168.1.1] port = 10296 group = sales [192.168.1.2] port = 10345 group = sales [192.168.1.3] port = 19002 group = technical In such a case as above, onlye the first two machines (192.168.1.1 and 192.168.1.2) will be notified. All you need configured on the machines that need to receive these messages is software like YAC (Yet Another Callerid program) which you may get from http://sunflowerhead.com/software/yac/ You will only need to configure the broadcast application to connect to the right port. The usage and testing informtion is quite well documented in the accompanying README file. Hope you find it useful! Cheers, Gerald. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Application: Broadcast
Hello, I've released an Asterisk application under the terms of the GNU GPL. You may find it here: http://psg.com/~begg/projects/ A short exerpt from the README: -- Broadcast is an Asterisk (http://www.asterisk.org) application which you may use to send a generic message over TCP/IP to any number of computers running software configured to listen for these types of messages. Being written in C, Broadcast will be dynamically loaded onto the Asterisk program on startup, making it a highly reliable and scalable option when compared with other solutions based on the Asterisk Gateway Interface (AGI) system... -- Hope someone finds it useful! Cheers, Gerald. PS: Sorry for the cross posts! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Getting phpconfig to work?
Hi, When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; It's quite likely that your Apache+PHP installation is incomplete / broken. You may want to check that out. Cheers, Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wctdm and two tdm cards
If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. To completely reset the TDM cards before they can be reliably detected again, you may have to completely power down the machine - even to the extent of pulling out the power plug and replacing it, then booting up. Regards, Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Card Problems
Hi Mark, On Mon, 28 Feb 2005, Mark Kidd wrote: modprobe zaptel - no problems [EMAIL PROTECTED] root]# modprobe wcfxo I'm just curious, did 'modprobe wcfxo' ever work? I seem to remember that for the TDM400P suite, the module to load was (rather confusingly) 'wcfxs', even though you've got FXO modules on the card. we are running the 4 port fxo digium card. so normal the modprobe wcfxs no problems modules load and board comes up after starting asterisk. That's TDM04B, right? If you don't have the Wildcard X100P (or something of the sort) plugged too then I see no reason to be loading 'wcfxo'. Hope that helps. Regards, Gerald. PS: The module name was later changed from 'wcfxs' to 'wctdm' (to avoid confusion I think. So, if you have no X100P, I think you can safely ignore loading 'wcfxo') ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPCB
On Thu, 24 Feb 2005, HASS, JOHN wrote: type=peer For some reason type=friend seemed to solve a similar problem I had (not with IP Clearing Board, though). I was kinda too busy to figure out why it solved the problem, actually [sorry] but it *may* be worth checking out. Then, just to clarify, that section in your sip.conf seems to suggest that you've only configured your server to allow calls to be terminated *from* IPCB? I.e the IPCB registering with your Asterisk server. Perhaps you might want to think of a register statement? I admit am not completely familiar with the way they get things running though, just a guess. Hope that helps. Regards, Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to manage Digium TDM04B outgoing calls correctly
So if you think the server can handle 5 TDM400P cards let me know. I've done an installation with 5 TDM400P cards - 4 PSTN lines and 12 analog phones. There are no outstanding issues that havent been solved by tweaking a particular config option (e.g echo, callprogress issues etc...). Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Yup, I found their support very unhelpful and unwilling to go the extra (or even the first) mile.. Might ACPI (not APIC) have anything to do with this condition? I once had a hard time with a bunch of cards which were not taking interrupts. I disabled ACPI interrupt routing (from the grub boot prompt, put pci=noacpi) and everything started working. Well, these were TDM400P cards (5 of them) anyway with a different type of machine altogether but it just might be worth checking out. Rgds, Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM31B has no interrupts?
Hi, I've installed a TDM31B card successfully but had a few problems making calls through it - summary is below: o Calls cannot be placed using an analog phone o The interrupts count value in /proc/interrupts remains at zero (see below) CPU0 0: 7495 XT-PIC timer 1: 7 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 2 XT-PIC rtc 9: 0 XT-PIC acpi 11:508 XT-PIC eth0 12: 0 -- XT-PIC wctdm -- 14: 2662 XT-PIC ide0 NMI: 0 LOC: 0 ERR: 0 MIS: 0 o I've tried this card in all three PCI slots but no luck o I've tried two other TDM31Bs in a similar manner with no luck o I've tried the same with a TDM22B and get similar behaviour Could all my PCI slots be dead or is it likely that all 3 TDM31B cards are dead + the TDM22B? Any clues are highly appreciate. Rgds, Gerald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM31B has no interrupts?
Hi, Thanks for taking time to answer. Not enough info in the above to hint at the problem. What linux distro, SuSE Linux 8.2 2.4.20-4GB what does your /etc/zaptel look like, For the TDM22B card: fxoks=1-2 fxsks=3-4 loadzone = uk defaultzone=uk zapata.conf signalling=fxo_ks echocancel=yes echocancelwhenbridged=yes echotraining=400 callprogress=yes busydetect=1 busycount=7 relaxdtmf=yes channel = 1,2 signalling=fxs_ks group=2 context=incoming channel= 3,4 , what steps did you take to start the drivers (eg, modprobe, ztcfg), etc. linux:~ # modprobe zaptel Warning: loading /lib/modules/2.4.20-4GB/misc/zaptel.o will taint the kernel: no license See http://www.tux.org/lkml/#export-tainted for information about tainted modules Module zaptel loaded, with warnings linux:~ # modprobe wctdm Warning: loading /lib/modules/2.4.20-4GB/misc/wctdm.o will taint the kernel: no license See http://www.tux.org/lkml/#export-tainted for information about tainted modules Module wctdm loaded, with warnings linux:~ # ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. linux:~ # Since we're forced to guess at that stuff, here's a list of things that might have an impact. - ensure the definitions in /etc/zaptel.conf are reasonable Hope they are. - run ztcfg -vvv from the command line. Any errors? No errors as above. - run 'modprobe wctdm', any errors? The only message that shows is a warning about the kernel being tainted. I checked the link I was referred to and it should really have no effect on the operation of the card (s). - what does dmesg, lspci, and lsmod output say? dmesg: Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) Registered tone zone 4 (United Kingdom) Registered tone zone 4 (United Kingdom) lspci: 00:00.0 Host bridge: VIA Technologies, Inc. P4M266 Host Bridge 00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP] 00:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus Master IDE (rev 06) 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97 Audio Controller (rev 50) 00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 74) 01:00.0 VGA compatible controller: S3 Inc. [ProSavageDDR K4M266] lsmod: Module Size Used byTainted: P wctdm 25568 0 (unused) zaptel183616 0 [wctdm] snd-pcm-oss45888 0 (autoclean) snd-mixer-oss 13560 0 (autoclean) [snd-pcm-oss] isa-pnp29672 0 (unused) ipv6 134388 -1 (autoclean) raw139414516 0 (unused) ieee1394 32880 0 [raw1394] via-rhine 12176 1 mii 2304 0 [via-rhine] snd-via823312516 0 snd-pcm62912 0 [snd-pcm-oss snd-via8233] snd-timer 11904 0 [snd-pcm] snd-ac97-codec 31152 0 [snd-via8233] snd-mpu401-uart 3360 0 [snd-via8233] snd-rawmidi13824 0 [snd-mpu401-uart] snd-seq-device 4000 0 [snd-rawmidi] snd35940 0 [snd-pcm-oss snd-mixer-oss snd-via8233 snd-pcm snd-timer snd-ac97-codec snd-mpu401-uart snd-rawmidi snd-seq-device] soundcore 3396 0 [snd] reiserfs 200532 1 - what does zttool show? I realized I need the package libnewt to get this to compile. Meanwhile does the above information reveal anything? Is there any BIOS setting I need to tweak? Thanks in advance. Rgds, Gerald. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM31B Interrupt Issue SOLVED! :-)
-- My apologies if this gets posted twice. I made a mistake with my from address. -- -- Hi All, Many thanks to everyone that gave input on the above issue. I'm glad to announce its been solved. The trick: -- TURN OFF ACPI! -- With SuSE you can do this by setting the boot option pci=noacpi. Everything now works flawlessly except for some suspicious static that I heard on one of the 3 TDM31B cards, which vanished after I reloaded the modules. Incidentally the technical reference booklet that came with the PC says the slots are PCI ver 2.1 compliant. I almost thought that would be a problem. Well, it turns out it isnt after all. Thanks again. In case anyone is interested, I've included my scratch notes on what I went through with this. Rgds, Gerald. --[Use at your own risk!]-- - Download asterisk, zaptel from CVS - Hack zaptel.c and wctdm.c modules to have the kernel_version string in them. This will allow them to be loaded (had to do this for SuSE :-(). - Edit /etc/zaptel.conf to tell the signalling, zone etc... - Edit /etc/asterisk/zapata.conf to reiterate this stuff for asterisk - Run ztcfg -vvv and note the output HARDWARE WOES - Disable xwindows - Disable USB (remove /etc/hotplug/usb.rc or rename it) - Tweak BIOS IRQ stuff THE PCI SLOTS SHOULD BE PCI 2.2 COMPLIANT! TURN OFF ACPI I.E WHEN BOOTING USE pci=noacpi -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 - TDM31B Card Installation Difficulty
Hi, o I purchased 3 TDM31B cards and fixed them in my computer (in 3 PCI slots) o I downloaded the latest Zaptel source from CVS, compiled it and loaded modules zaptel.o and wctdm.o. o I successfully configured them from /etc/zaptel.conf as shown in the information below. ztcfg returned no errors - see the report below. o I successfully configured /etc/asterisk/zapata.conf (see info below). o I configured an X-Lite phone to test with an analog phone plugged into one of the channels. The problems are: o I cannot make a call from the analog phone (Saachi phone, KX-T3223) connected to one of the FXS ports. When I pick up the receiver, I hear the dialtone but when I press the buttons, asterisk seems not to get the numbers dialled, both using pulse and touch tone dialling. o I can call the analog phone from X-Lite however on receiving, I cannot hear much voice. What I hear is choppy sound corresponding to whatever I say from the analog side. When someone speaks from the X-lite side, nothing is heard from the analog phone. o There are three FXS ports where there is no dialtone - but the phone is actually powered - I can hear touch tone / pulse when I dial. o There are three FXS ports that give neither power nor dialtone. What could the problem be? Any help will be highly appreciated. Please find below abit of information I thought may be useful. Please let me know if more is needed. EXTRA INFORMATION - linux:/usr/src/new # uname -a Linux linux 2.4.20-4GB #1 Mon Mar 17 17:54:44 UTC 2003 i686 unknown unknown GNU/Linux linux:/usr/src/new # linux:/usr/src/new # ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Loopstart (Default) (Slaves: 01) Channel 02: FXO Loopstart (Default) (Slaves: 02) Channel 03: FXO Loopstart (Default) (Slaves: 03) Channel 04: FXS Loopstart (Default) (Slaves: 04) Channel 05: FXO Loopstart (Default) (Slaves: 05) Channel 06: FXO Loopstart (Default) (Slaves: 06) Channel 07: FXO Loopstart (Default) (Slaves: 07) Channel 08: FXS Loopstart (Default) (Slaves: 08) Channel 09: FXO Loopstart (Default) (Slaves: 09) Channel 10: FXO Loopstart (Default) (Slaves: 10) Channel 11: FXO Loopstart (Default) (Slaves: 11) Channel 12: FXS Loopstart (Default) (Slaves: 12) 12 channels configured. linux:/usr/src/new # /etc/zaptel.conf: fxols=1-3 fxols=5-7 fxols=9-11 fxsls=4,8,12 loadzone = us defaultzone=us /etc/asterisk/zapata.conf: [channels] signalling=fxo_ls echocancel=16 echocancelwhenbridged=yes is in milliseconds pulsedial=yes group=1 context=default callprogress=yes busydetect=1 busycount=7 relaxdtmf=yes channel = 9-11 channel = 1-3 channel = 5-7 signalling=fxs_ls group=2 context=incoming channel= 4,8,12 Gerald. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 - TDM31B Card Installation Difficulty
Hi Steve, If you call from X-Lite to the demo menus can you hear them clearly (no choppy sound)? Actually I can't - the sound is still choppy! Interesting. When I unload the zaptel and wctdm modules the problem goes away (I can hear the demo files quite clearly from the X-Lite phone). Given the problems you are having this might point to a bad TDM100P card. Mmm. I have a spare one. I'll replace the one that doesn't give dialtone and see what happens. Thanks alot Steve. I'll fix the card and let you know what happens. Rgds, Gerald. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Out through Provider with Authentication
On Thu, 9 Sep 2004, Karl Brose wrote: In order to dial out to a sip provider, you need to configure that provider in your sip.conf file as a peer with your proper username and secret, etc. Cool! Just found that in the handbook too a second or two ago :-) Thanks for taking time to answer this. Three Cheers! Gerald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Out through Provider with Authentication
On Fri, 10 Sep 2004, Johannes Hollerer wrote:
I tried to make a call to extension 2001 with the setting
[EMAIL PROTECTED] (Detailed: exten =
_7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1}) which does not work at
all - i always get the failure message: No such host
provider.com/2001 (the number i dialed) - why ??
What I understood from Karl's message is that you need to create a peer in
sip.conf. For example below:
-- sip.conf --
[myprovider]
type=peer
username=USERNAME
host=PROVIDER.COM
secret=SECRET
--
Then in extensions.conf, do the following:
--
exten = _7.,2,Dial(SIP/myprovider/${EXTEN:1})
--
This should work. What Karl meant is that using the statement below:
--
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED])
--
Will only work if you are dialilng a *specific* extension on provider.com.
The statement below:
--
exten = _7.,2,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
--
Is illegal.
Cheers,
Gerald
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Re: [Asterisk-Users] Asterisk testbed for teaching connecting to a PRI-ISDN
On Fri, 10 Sep 2004, Francesco Delfino wrote: [...]One of the box will represent the Telco, the other two, the two companies PBX. I would like to know if it is needed something between the point-point connections or it is possible to just cross-connect them. As more experienced people prepare to reply, I'd like to give my [highly theoretical] opinion (I'm still waiting for hardware I ordered): I think it is possible to just cross connect them, as long as you get the signaling right. In my opinion, the Box simulating the telco should signal as the network side and the one representing the company should signal as the customer side... Hope that makes sense. Cheers, Gerald. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Out through Provider with Authentication
On Thu, 9 Sep 2004, Johannes Hollerer wrote:
I try to dial out through a Provider, but for that i need to be
authenticated - it actually does not work !.
For my tests I did not need to be authenticated. This is what I used in
asterisk:
exten = _7.,2,Dial(SIP/PROVIDER.COM/${EXTEN:1})
When I tried to use your scenario, as below
exten = _7.,2,Dial(SIP/USERNAME:[EMAIL PROTECTED]/${EXTEN:1})
Here's what I get in my logs:
Sep 9 18:10:56 WARNING[137570304]: chan_sip.c:902 create_addr: No such
host: PROVIDER.COM/72312
What I gather from this is that its not legal to Dial() like that. In my
limited SIP knowledge, it makes sense - you do not need to have a username
and / or password to place calls to extensions that a given provider (e.g
PROVIDER.COM) serves - if they do not serve those extensions, they will
give a 404 Not Found error.
Hope that helps...
Gerald.
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Re: [Asterisk-Users] Dialing Out through Provider with Authentication
But the provider also has a gateway to provide the possibility to
call to the pstn (and the pstn number exists) - so what i tried to
achive is to call an external pstn number thru that gateway. This
works if i connect the xlite client directly to the provider - then
i can dial the external number.
Alright, I see what you mean. Have you entered a register statement in
sip.conf, then? I.e something like
register = USER:[EMAIL PROTECTED]/EXTENSION
What I understand is that this will result in your Asterisk Server
registering on that provider's server as one of its users. Now the
question is how you dial out through this registration... Ya? That much I
don't claim to know. I just hope guys who have done this are reading this
thread. I'd like to learn this too.
However try using the dial below with the above register statement in
place (don't forget to reload your Asterisk server).
exten = _7.,2,Dial(SIP/PROVIDER.COM/${EXTEN:1})
Ideas, anyone else?
Gerald
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Re: [Asterisk-Users] Placing Asterisk between existing PBX and PSTN
Hi All, Just wondering if anyone could have by chance taken a look at the scenario below... I checked up http://www.asteriskpbx.com/index.php?menu=support and it looks like Asterisk-Users is the correct list to post this (I think...). I'd really appreciate any insight. Gerald. On Mon, 6 Sep 2004, Begumisa Gerald M wrote: Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup --- PSTN +---+ --|| ||--A1 --|| PBX ||--A1 --|| ||--A1 --|| ||--A1 +---+ A1,A2,A3,A4 are analog extensions Setup With Asterisk --- PSTN +--+ +---+ --||| |||| ||--A1 --|FXO Card|| Asterisk ||FXS Card|| PBX ||--A2 --||| |||| ||--A3 --||| |||| ||--A4 +--+ +---+ So they only pay for the Asterisk box, the FXO Card (TDM04B) and FXS Card (TDM40B). I'd appreciate any yes/no/been there answers. I just want to make sure about this, in case there's anyone that's done this before. Thanks in advance. Gerald. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Placing Asterisk between existing PBX and PSTN
Hi Jens, Thanks alot for your input, I do appreciate it! [...] I would like to suggest that you don't try this with analogue lines (fxo) and extensions (fxs) - you will not be able to monitor call progress and lose all (possible) DDI information. Imagine my original setup was purely analog i.e I have 4 analog lines from the local telecom company. If I plugged these lines into the Asterisk FXO card and then plugged the PBX into the Asterisk FXS card, I'm thinking I would be able to use the calling card in addition to making VoIP calls application if for example I set up an extension context as below: [analogextensions] exten = 101,1,CallingCard exten = 101,1,Congestion exten = 102,1,Dial(SIP/[EMAIL PROTECTED]) I.e if anyone on the analog phones dials 101, they get the Calling Card application, if they dial 102, they get connected to some SIP phone somewhere etc... Would this minimum functionality work? Thanks! Gerald. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Placing Asterisk between existing PBX and PSTN
Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup --- PSTN +---+ --|| ||--A1 --|| PBX ||--A1 --|| ||--A1 --|| ||--A1 +---+ A1,A2,A3,A4 are analog extensions Setup With Asterisk --- PSTN +--+ +---+ --||| |||| ||--A1 --|FXO Card|| Asterisk ||FXS Card|| PBX ||--A2 --||| |||| ||--A3 --||| |||| ||--A4 +--+ +---+ So they only pay for the Asterisk box, the FXO Card (TDM04B) and FXS Card (TDM40B). I'd appreciate any yes/no/been there answers. I just want to make sure about this, in case there's anyone that's done this before. Thanks in advance. Gerald. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
