Re: [Asterisk-Users] POE hub
Dean Collins wrote: Yep, POE has turned out to be a real fizzer. Whilst a great idea for Access Points (particularly ceiling mounted AP's They are *far* more useful for simplifying phone wiring. so you don't need to run power points) but apart from that the whole concept has just died. Not really. People are just stuggling with the cost. They specified something rather complex, and the MCUs and other stuff needed to support that complexity will stop PoE ever being nick and dime stuff. Single chips to do the power control have only recently become plentiful. That should make the cost a bit more reasonable, and probably make it tolerable over the next year. Don't expect to get a $25 8 port switches with PoE any time soon, though. Regards, Steve ___ 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] AGI - How to Make Calls and Bridge to Original Incoming
TC wrote: Why not just keep it simple use dial with Macro argument and this std macro-screen like this http://lists.digium.com/pipermail/asterisk-users/2005-March/098257.html Thank you so much! I was not familiar with this option since we only run STABLE and this feature is only available from CVS. Since we cannot risk running our main switch on CVS what I have done is set up a second server in the rack running the current CVS version, IAX calls from our production server to this box which avails itself of this new Dial macro feature, uses IAX to link back to the production server to make the outbound calls, and then bridges the call once an outbound call has been accepted. Thanks again for your suggestion. Now to wait for Dial with Macros to make it to stable :-) g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ 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: Writing To Multiple MySql Tables
Is there any other way to connect multiple tables and fields to read and write in the dialplan? (simple inserts queries). Perhaps via app_dbodbc or res_sqlite? Rafal -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.10 - Release Date: 13/05/2005 ___ 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] zttest
After I run it, I get the following: 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% Just for reference, I'm running a PIII-800Mhz and I get (with no particular load on CPU) -Best: 100.00 -- Worst: 99.987793 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 99.987793% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% ___ 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] Callerid on PC and more
I suppose by this you mean some sort of client software installed on the client PC that listens to events targeted at a particular port this software is listening to. If this is the case, how do you make Asterisk communicate with this client software? I use yac and system() with the nc comand The only unfortunate thing is having to issue one system() per LAN or external ip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 117
Date: Sun, 15 May 2005 15:17:53 -0700 From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 911 Options To: Ira Burton [EMAIL PROTECTED], Asterisk Users Mailing List - On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote: I am curious if anybody has pointers on the best way to get the 7 digit PSAP number for an area. I am thinking about making a '911' extension that will dial the PSAP number, wait for the PSAP to answer and play a message giving the address of the originating call, and replay the the information every three minutes. I am concerned what may happen if my children try to dial 911 in an emergency but do not yet know our address. You can buy them on CD, however to do E911 you have to have a special trunk to the switch that the PSAP is off of, which transmits the E parts of E911 not just the audio. Where to buy them I dont know offhand, I do specifically recall seeing pages that sold national CDs (how adt, onstar, even other PSAPs contact a specific PSAP when needed). You can buy them from a few different vendors. Last I checked it was like $50,000. You can also just call up your state administrators for PSAP. Keep in mind that numbers can change, though, which is why updates are important. Justin Newman Newman Telecom, Inc. ___ 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] A hook flash sent using RTP for telephony signals (RFC2833) does not flash zap channel
I just registered ID 0004283 at http://bugs.digium.com for the problem described in subject (found when using a Linksys PAP2-NA). I don't know where the proper forum is to discuss, so I'm hoping anyone interested will read the bug and let me know your thoughts, either at bugs.digium.com, here, or by emailing me directly (or, please suggest another forum that is more appropriate). As an aside, if you know how to make a Cisco 7960 running SIP send a flash command (SIP, RTP, or otherwise), I'd really like to know. /** Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo, Ventura, Los Angeles, Orange T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo ***/ ___ 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] 2 minutes pause before ring on H323 channel
John Daragon wrote: Yep - down in openh323/src/transports.cxx there's a method H323TransportAddress::GetIpAndPorts() which is called (eventually) by MakeCallLocked(). This in turn calls GetPortByService() and GetHostByAddress(). My guess is that the 60 second wait is caused by a request to a DNS server that is never honoured. Of course, I've been wrong before... Hi John, Thank you for the help ... i'll check if it is DNS timeout later this week when i have time to play with it and i'll post result here. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SpanDSP TXFax and multipage faxes problems (aditional info)
OK I see the ponit (although I never said that second page is interrupted - I said that in some combinations of resolutions and TIFF options receiving fax spits another blank sheet of paper beside the clearly received first page). I have read someware (some faxing tutuorial) that there is some kind of control code (6 EOLs I belive) that should be in TIFFs between pages and that is called RTC (return to control). Whole thing wasn't explained completly, and since I have no access to adequate specifications, I have thought that problem is there (I mean I thought that there is some negotiation before every page sent). Anyhow, as a workaround, I will modify faxing program I have made for secretary, to sent one page at the time (so multi page faxes are sent in multiply calls to a receiving fax). Regards, Nenad ___ 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] Callerid on PC and more
How do you make yac open a webpage?? Or what are you doing with yac on the client pc? Is there any way to configure yac with a diff. skin or something? Or plain old black small screen is ugly :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Wilson Pickett |Sent: Lunes, 16 de Mayo de 2005 01:50 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Callerid on PC and more | | I suppose by this you mean some sort of client software installed on | the client PC that listens to events targeted at a particular port | this software is listening to. If this is the case, how do you make | Asterisk communicate with this client software? | |I use yac and system() with the nc comand | |The only unfortunate thing is having to issue one system() per |LAN or external ip. |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zttest
Hi Waldo, I would be money on your problem being related to the accuracy of zttest. One way of checking IRQ's is to run cat /proc/interrupts, but it is a lot more accurate to run lspci -v and lspci -vb. I would recommend Googling the lspci command, although the output is pretty self explanatory. The TDM appears as a TigerJet card, not sure what TE410P will list as. PCI devices have their IRQ's dictated by the BIOS of the host system. How (and if) you can configure these manually depends on the type of BIOS you have... in our IBM xSeries 206 we had to actually juggle cards between slots to get it to assign a unique IRQ to the TDM400P. Good luck! D. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Waldo Rubinstein wrote: Damian, Thanks for your input. Hyperthreading is in fact enabled and now that you mention this I will disable it. The reason I ask is because under some load (may be 40 simultaneous calls), voice quality degrades. We have audio problems where one party hears the other but not viceversa and then it all works fine. It's random audio quality problems in general. During these cases, I'm constantly running vmstat 1 and CPU utilization is always 85%+ idle. I will also look into setting the TE410P in its own IRQ. Do you know how I can do that? Is that a motherboard BIOS setting or is it something that needs to be done to the TE410P itself? Thanks, Waldo On May 16, 2005, at 12:59 AM, Damian Funnell wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). Do you have issues with your * box? If so then I would start worrying about zttest output (and thinking about disabling hyper threading on those dual Xeons), otherwise have a smile and a beer and pity us poor fools who have had problems due to poor results. Cheers, Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Waldo Rubinstein wrote: I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% --- Results after 57 passes --- Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793 What does this mean? Should I have expected to get 100% across the board? This is from a TE410P running on Debian 2.6.11-1-686-smp on a dual Xeon 2.4GHz server. Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Number Portability Details
Hi, I'm seeking to change my service provider (after ten months, I've had it with broadvoice), but I would like to keep my 310 number. I've been digging through the lists of other providers and am considering telasip (good plans and support number transfers). My concern is what precisely happens when a number is transferred from one service provider to another. After the transfer is complete, when someone dials my number, will it go to broadvoice's servers/routers initially, and get bounced over to telasip? This would be much less than ideal, because I would not be escaping my outage problem. Or does a transferred line truly go straight to telasip, never seeing broadvoice's servers again. This is the solution I want, but I don't know how it works. The reason I ask is because I know with cell phones, having a number transferred results in a situation where the number initially terminates with the old provider, but then gets bounced (which can and does cause horrible call routing problems). I'd like to know what happens in this case before I pay for a number transfer that won't do me any good. Thanks in advance, Paul ___ 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] res_config_mysql.so relocation error
Hi, in my attempt to install ISDN BRI card, I loaded asterisk-addons. I think I went to fast and buggerd up the locations of the files and directories. cant load asterisk again, getting: [res_config_mysql.so] = (MySQL RealTime Configuration Driver) asterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_config_load Ouch ... error while writing audio data: : Broken pipe To revert; it should be ok for me to do cvs update on zaptel libri asterisk only? Or how can I repair my cause of relocation error? pse anyone rgds ___ 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] Callerid on PC and more
How do you make yac open a webpage?? Don't know, since I'm not triying to open a webpage Or what are you doing with yac on the client pc? The CID info pops up so someone working on their PC can see who's calling. Especially nice for people with older phones that don't have CID at all. Is there any way to configure yac with a diff. skin or something? Or plain old black small screen is ugly :) you might read the info yac comes with and play with the source. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Gerenal settings conufsion
Personally, I'd like to see this changed so there are two 'general' sections--one for default parameters to use unless overridden when there *is* a peer section below, and a different one to describe parameters to use when the remote peer is not previously known. I know there are ways to accomplish this with the existing sip.conf structure but it seems very counter-intuitive. In CVS head you can use the new templates for peers with a section, thus leaving the [general] settings to unknowns. You have to set ALL the parameters though, since the channel has it's own defaults embedded. Example: If you don't have a context= setting in general, we will use default as the default setting for inbound calls from unknown. THis is the way asterisk behaves in quite a lot of places in regards to contexts. So if you do want those calls to be going into a black hole, you will have to create a black hole context and set it in sip.conf. Regards, /Olle ___ 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] mISDN error while compiling
Hi, From the chan_misdn readme: Now I use Kernels 2.6.9 and it works perfect. with kernels = 2.6.10 there is a very litle bug in hfc_multi.c which causes the module not to compile, it can be easyly fixed by changenging pci_findsubsys to pci_getsubsys in code. Hope this helps ___ 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] Asterisk@home 1.0 + Sipgate UK/SIP Provider
Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of 8|. to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk console returns -- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack -- Called sipgate/## But the call is never made, and no errors reported. I am behind a router (ipcop) but I would have thought I dont need to set any ports as its outgoing, and there is no outgoing blocks on the router. Edit SIP Trunk -- Outbound Caller ID: my sip number Dial Rules: 8|. Trunk Name: sipgate PEER Details host=217.10.79.219 secret=** type=peer username=### Sipgate username Edit Route --- Dial Patterns: 8|. Trunk Sequence: SIP/sipgate Thanks for your time. ___ 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] AreskiCC
Hi, I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and Postgres 7.4. First of all the instructions are very confusing and hard to follow if you are not an expert. But, I managed to install it andobviously t doesnt work. The other instructions I found on wiki are a great effort but incomplete. Basically the first thing that happens is that when I load /areskicc/Public/index.php it refuses my username and passwork (AUTHENTICATION REFUSED, please check your login/password! ) which I guess is the same as the one I configured on defines.php right?) and after I reinsert it I get the error: Method Not Allowed. The requested method POST is not allowed for the URL /areskicc/Public/index2.php. In any case, does anybody know of any better instructions on how to install and configure AreskiCC? ___ 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] callback problem
hello i am trying to make a callback solution. client will call callback number and call is terminated. now callback server will create a call for that client. actually i have a problem in this process. that server is creating call to client (UA) when previous call is not disconnected yet. UA--Asterisk(callbacknumber) callis answered UA--Asterisk(callbackserver) call is created when previous call is not hangup any solution how to resolve this problem i am stuk here can any one help me in this Kamran __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ 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] SIP--h323 conversion
Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sipASTERISKh323-GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could someone help me with configuration of Asterisk? I installed [EMAIL PROTECTED] 1.0 and oh323 0.6.5 Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel.conf in /etc not /etc/asterisk - historical reason?
Hello all I am in the process of trying to create a more fault tolerent HW setup for my asterisk platform, its all going well and I intend to do a wiki about it once its seen to be working. One thing gets me, and hopefully someone here can confirm my suspision - why is zaptel.conf not with the other asterisk files (I assume it is because its responsable for bringing up the hardware, not strictly part of the asterisk application) Would someone care to confirm my suspision, and if I'm wrong advise me why. As a follow on to this - if i were to move it somewhere else, is it the somthing.c file that would need to be changed to reflect this move. Thanks David ___ 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] Asterisk@home 1.0 + Sipgate UK/SIP Provider
Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of 8|. to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk console returns -- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack -- Called sipgate/## But the call is never made, and no errors reported. I am behind a router (ipcop) but I would have thought I dont need to set any ports as its outgoing, and there is no outgoing blocks on the router. Edit SIP Trunk -- Outbound Caller ID: my sip number Dial Rules: 8|. Trunk Name: sipgate PEER Details host=3D217.10.79.219 secret=3D** type=3Dpeer username=3D### Sipgate username Edit Route --- Dial Patterns: 8|. Trunk Sequence: SIP/sipgate Thanks for your time. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP--h323 conversion
This is relatively straight forward, you can either use Nufones Implementation or the OH323 package. Both work relatively well. However, I've had issues presenting a GateKeeper ID from Asterisk to carriers that authenticate based on that in the past. Regards, Sahil Gupta VoiceValley On Mon, 16 May 2005, Micko wrote: Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sipASTERISKh323-GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could someone help me with configuration of Asterisk? I installed [EMAIL PROTECTED] 1.0 and oh323 0.6.5 Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POE hub
Steve Maroney wrote: The cheapest I have found was a 3COM 24 Port for $799.00. Thank you, Steve Maroney Be warned, we are a 3Com house, and I ordered a 4400 PWR to test it would work with our Siemens hard phones. Lucky I did, because it turns out they are not compatible! It seems the 3Com POE switches will only power 3Com devices. Instead, I ordered a bunch of PowerDsine injectors which work fine, and power a much greater range of devices. Regards -- Chris Hills IT Services North East Worcestershire College ___ 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] POE hub
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=51268item=5774375303 rd=1ssPageName=WDVW#MyDescription I found these on Ebay, what do you think? They are certainly cheap enough. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] chan_misdn and passive BRI cards
Has anyone got chan_misdn working with passive BRI cards yet? I've tried both hfc (hfcpci.ko) and w6692 (w6692pci.ko) cards, but when I start asterisk I get the following when chan_misdn is loaded: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) == Registered application 'misdn_set_opt' debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 Init. Stack on port:1 TE Stack No Upper ID port:1 init_stack: Success Ouch ... error while writing audio data: : Broken pipe After all the effort of recompiling my kernel (2.6.11 and 2.6.8) for misdn I'd really like to get asterisk working with misdn. ___ 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] pickup timeout
Hello, I am looking for how to increase the pickup timeout. If a call is not picked up in 20 seconds asterisk automatically hang it up indicating the message: Nobody picked up in 2 ms How can I increase this timeout? Thank you very much. Regards, Alberto ___ 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] Asterisk@home 1.0 + Sipgate UK/SIP Provider
Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of 8|. to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk console returns -- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack -- Called sipgate/## But the call is never made, and no errors reported. I am behind a router (ipcop) but I would have thought I dont need to set any ports as its outgoing, and there is no outgoing blocks on the router. Edit SIP Trunk -- Outbound Caller ID: my sip number Dial Rules: 8|. Trunk Name: sipgate PEER Details host=217.10.79.219 secret=** type=peer username=### Sipgate username Edit Route --- Dial Patterns: 8|. Trunk Sequence: SIP/sipgate Thanks for your time. ___ 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] FXO/FXS suggestions:
Im looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. Your only choice for zaptel type is the TDM card. Probably the next best choice is the spa3000. I wish I could find out for sure how well the spa-3000 FXO works with * and same for the Grandstream Handytone 488. I need FXO-SIP conversion in places where I don't need a PC. The spa3k works pretty good in most cases; been using one for several months without any major issues and know of lots of others doing the same. It can be a little time consuming figuring out the many config options available though. (Use the tool at voxilla.com to help, and then review what the tool changed as sort of an educational thing.) Its my understanding from others the 488 does not support the fxo port as an addressable sip port from *. Its only accessible from the fxs port, apparently as a fallback or something like that. I've got my spa3k set up sort of like a mini pbx. The dialplan within it is configured to send all calls starting with an 8 out the voip port, while all other calls default to fxs - fxo dialing. No remedial training required for non-technical users, and * is not in the middle of normal local calls. Another spa3k is configured for the fxs port to register with *, and the fxo port also registers with asterisk, forcing asterisk to be in the middle of all calls. In this case, one must use the g711 codec as apparently there isn't enough horsepower to run to simultanous g729 sessions at the same time. For this config to function, one has to essentially call-forward any incoming fxo calls to asterisk. Not a big deal at all; works fine. Given the cost of the spa, try one to see if it meets all of your requirements. ___ 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] POE hub
I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Hope you really meant a cheap switch... you don't want to use hubs of any sort in the asterisk environment since hubs are limited to half-duplex ethernet connections. Check the NetGear stuff. I believe someone indicated they are selling a poe switch for something like $400. ___ 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] skype channel
I'm not overly familiar with the Skype API. Last I heard the API is missing the necessary features to make a full client, this was obviously done on purpose by Skype. I think there are some solutions to get a third party tool to run along with Skype. On 5/15/05, Wessel de Roode [EMAIL PROTECTED] wrote: I've just added a view day's ago some information on it on the wiki. As far as I know there is nothing really working 'yet' but I'm sure since the API is out it' won't take long :-) http://www.voip-info.org/tiki-index.php?page=bounty%20skype Wessel de Roode -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zaptel.conf in /etc not /etc/asterisk - historical reason?
In article [EMAIL PROTECTED], David John Walsh [EMAIL PROTECTED] wrote: One thing gets me, and hopefully someone here can confirm my suspision - why is zaptel.conf not with the other asterisk files (I assume it is because its responsable for bringing up the hardware, not strictly part of the asterisk application) Yes. Zaptel came before Asterisk and is independent of it. It is possible for other non-Asterisk software to make use of Zaptel, without Asterisk needing to be present at all. Would someone care to confirm my suspision, and if I'm wrong advise me why. As a follow on to this - if i were to move it somewhere else, is it the somthing.c file that would need to be changed to reflect this move. Don't know, but I have trouble understanding the need to move it. The only place that it would make sense to move it to would be /etc/zaptel/zaptel.conf, but since it is a single file, why bother? It certainly doesn't belong in the /etc/asterisk directory. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] POE hub
Single port 3com injectors are really cheap. Like $20 a piece. Granted no one wants to have a MASS of POE injectors. For small 8 installations it might be manageable though. I haven't tried them with things outside of my 3com NJack.. I'll have to test it on the Polycom before I buy more. On 5/16/05, Chris Hills [EMAIL PROTECTED] wrote: Steve Maroney wrote: The cheapest I have found was a 3COM 24 Port for $799.00. Thank you, Steve Maroney Be warned, we are a 3Com house, and I ordered a 4400 PWR to test it would work with our Siemens hard phones. Lucky I did, because it turns out they are not compatible! It seems the 3Com POE switches will only power 3Com devices. Instead, I ordered a bunch of PowerDsine injectors which work fine, and power a much greater range of devices. Regards -- Chris Hills IT Services North East Worcestershire College ___ 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 -- Is it something someone said, was it something someone said? ___ 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] Always Ringing
Hi all, I am using chan_h323 from Asterisk CVS to interconnect with GNUGK v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on Asterisk. However, I only heard ringing when the call was answered on SIP side. Below is the debug from chan_h323. Any help is welcome. Thanks. *CLI == New H.323 Connection created. -- Setting up Call -- Call token: [ip$22.7.20.32:30012/16050] -- Calling party name: [6907] -- Calling party number: [6907] -- Called party name: [0069777] -- Called party number: [0069777] --Received SETUP message =-= In OnAnswerCall for call 16050 - Progress Indicator: 0 - Inserting PI of 0 into ALERTING message -- Started logical channel: sending G.729 -- channelsOpen = 1 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 22.7.20.32 -- remotePort: 51048 -- ExternalIpAddress: 0.0.0.0 -- ExternalPort: 17816 -- Started logical channel: receiving G.729 -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: ExternalRTPChannel Destroyed ExternalRTPChannel Destroyed -- Executing Dial(H323/ip$22.7.20.32:30012/16050, SIP/69777) in new stack -- Called 69777 -- SIP/69777-c6ce is ringing Sending alerting -- SIP/69777-c6ce answered H323/ip$22.7.20.32:30012/16050 Answering call ip$22.7.20.32:30012/16050 -- Transmitting RFC2833 on payload 96 -- Received Facility message... =-= In OnConnectionEstablished for call 16050 -- Connection Established with 6907 [22.7.20.32] -- Received Facility message... -- Started logical channel: receiving G.729 -- channelsOpen = 3 External RTP Session Starting RTP channel id 1 parameters: -- Received Facility message... -- Received RELEASE COMPLETE message... -- ClearCall: Request to clear call with token ip$22.7.20.32:30012/16050, cause EndedByRemoteUser -- Sending RELEASE COMPLETE channelsOpen = 2 channelsOpen = 1 channelsOpen = 0 ExternalRTPChannel Destroyed ExternalRTPChannel Destroyed ExternalRTPChannel Destroyed -- ClearCall: Request to clear call with token ip$22.7.20.32:30012/16050, cause EndedByTransportFail == Spawn extension (default, 0069777, 1) exited non-zero on 'H323/ip$22.7.20.32:30012/16050' -- 6907 [22.7.20.32] has cleared the call == H.323 Connection deleted. ___ 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] zttest
Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home 1.0 + Sipgate UK/SIP Provider
I am using Sipgate with [EMAIL PROTECTED] and this is how I have set mine up to have it working perfectly. Using the AMP Interface my trunk is setup as follows.. Under Trunk: Outbound caller ID is your full sip number including area code. Peer Detail: allow=ulaw authuser=539 (your sip number) canreinvite=no disallow=all dtmfmode=info fromdomain=sipgate.co.uk fromuser=539 (your sip number) host=sipgate.co.uk insecure=very nat=yes secret=XXX (your sip password) type=peer username=539 (your sip number) User Details: allow=ulaw authuser=539 (your sip number) context=ext-did disallow=all dtmfmode=info faxdetect=incoming fromdomain=sipgate.co.uk fromuser=539 (your sip number) host=sipgate.co.uk insecure=very secret=XXX (your sip password) username=539 (your sip number) User Context: Mine is ext-did Register String: 539:[EMAIL PROTECTED]/539 Hope this helps... Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Goodyear Sent: 16 May 2005 11:15 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] 1.0 + Sipgate UK/SIP Provider Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new [EMAIL PROTECTED] 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of 8|. to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk console returns -- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack -- Called sipgate/## But the call is never made, and no errors reported. I am behind a router (ipcop) but I would have thought I dont need to set any ports as its outgoing, and there is no outgoing blocks on the router. Edit SIP Trunk -- Outbound Caller ID: my sip number Dial Rules: 8|. Trunk Name: sipgate PEER Details host=217.10.79.219 secret=** type=peer username=### Sipgate username Edit Route --- Dial Patterns: 8|. Trunk Sequence: SIP/sipgate Thanks for your time. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] zttest
How do you disable hyper threading (what's the command and where is it placed)? Hyper-threading is a BIOS feature available on some Pentium 4 Xeon processors. If you have hyper-threading enabled your system may appear to have more processors than are physically in the system. Typically twice as many. You generally disable the hyperthreading feature through the BIOS setup program that's normally accessible when the system boots. ___ 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] POE hub
Rich Adamson wrote: I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Hope you really meant a cheap switch... you don't want to use hubs of any sort in the asterisk environment since hubs are limited to half-duplex ethernet connections. Check the NetGear stuff. I believe someone indicated they are selling a poe switch for something like $400. Can you actually buy hubs these days? With an 8 port switch costing $25, there doesn't seem much of a niche opportunity for a cheaper option. Regards, Steve ___ 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] zttest
On 06:37, Mon 16 May 05, Rich Adamson wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? You have to disable it in your server's BIOS. You can also try to install an uniprocessor kernel, but I don't know if that is enough. We had to disable HT too to get it all working the way we want. I think this is an issue with the HT support in kernel 2.4.X -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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] POE hub
Hope you really meant a cheap switch... Yes, I am going to use 16 port Linksys switches if I can't get POE units at a reasonable price. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] NAT and sip issues
I have an asterisk server behind NAT - no audio on the test external calls I have tried making so far. Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution evident from there, sounds like I have case 9. I would have thought that all I would have to do is port foward and have the external IP on the asterisk server, which I have done I have fowared 5060UDP, 8000UDP, and 35000 to 37000 UDP to the internal IP (192.168.1.115) I have put 35000 and 37000 into the rtp.conf as the start/end ports extracts of sip.conf: externip = 60.234.129.154 localnet = 192.168.1.115 localmask = 255.255.255.0 [88] type=friend secret=** dtmfmode=rfc2833 nat=yes host=dynamic canreinvite=no Trying with xlite at the other end Registered ok, can dial both ways, just no audio at all. In the log of xlite (cant see it at the moment as im not vnc'd in at the moment) it showed the xlite machines private IP address on some of the transactions that were logged. The client has a dynamic IP address so cant really be specified anywhere in the xlite configuration, I am also not sure on all the different firewall types. I was under the impression that there was no need to configure any portfowards at the sip softphone end. I will hopefully be using xlite or similar from a location with a very locked down firewall environment. I want to check all works on a normal nat router before trying it behind the nasty nat/firewall at this location. smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPS can now print and chartc
The latest version of IPSwitchBoard has been released: Version 0.116 - 16. may 2005 * Call Data Records can be charted (number of calls and duration of calls by the hour). * Hotel Application can now print the calls and charges * Many minor bug fixes FREE Download: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP--h323 conversion
Could you please give me an example of such configuration? Thank you! Regards On Monday 16 May 2005 12:28, Sahil Gupta wrote: This is relatively straight forward, you can either use Nufones Implementation or the OH323 package. Both work relatively well. However, I've had issues presenting a GateKeeper ID from Asterisk to carriers that authenticate based on that in the past. Regards, Sahil Gupta VoiceValley On Mon, 16 May 2005, Micko wrote: Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sipASTERISKh323-GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could someone help me with configuration of Asterisk? I installed [EMAIL PROTECTED] 1.0 and oh323 0.6.5 Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] cisco 3620 setup (newbie cisco alert)
I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots D-Channel Configurable modes Status E1 0/0 3115pri/channelized Administratively up E1 1/0 3115pri/channelized Administratively up I have set both controllers to hdb3, ccs with crc4 I notice that the D-Channel is set to 15. However, in the zaptel.conf file I use with my TE410p card, the D-Channel is set to 16. So, is cisco counting 0-14 (hence 15 is the D-Channek) and zaptel counting from 1-15 (and therefore 16 is the D-channel) ? Or is my config wrong on the cisco or zaptel ?? Any help would be welcome. Thanks ! Julian ___ 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] zttest
this was posted before: On 5/12/05, Colin Anderson [EMAIL PROTECTED] wrote: They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff --Digium PRI card setpci -v -s 01:04:0 latency_timer=ff --Digium 401 4 X FXS setpci -v -s XX:XX:X latency_timer=0 --1 entry for every other PCI card in system from LSPCI output, modify XX:XX accordingly Before setpci I would get best in ZTTEST at 99.987793% and worst ~ 99.95% After setpci best is 100% and worst is 99.987793% consitient. I use SpanDSP to recieve faxes and before faxes were garbled and now they are OK (BTW, now recieving ~150 faxes a day 99.95% OK, so SpanDSP *does* work fine, you just have to set it up right. Ask me how.) I put the setpci statements in /etc/rc.d/rc.local before my modprobes to the Digium hardware and Asterisk startup. I'm using a 4-way Netfinity FC2 * 1.0 stable I dunno, maybe the community is being too hard on Digium about the design of the card. I can understand their perpective, it's brutal to make a card that has to have such tight tolerances and make it work acceptably on the huge variation in white box hardware (or black box, in your case). There's a page on the Wiki about motherboards that work well with installation notes but that's pointless since motherboards are such a moving target. Even the motherboard vendor screwing around with BIOS updates can invalidate that information. Waldo Rubinstein escribi: I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% --- Results after 57 passes --- Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793 What does this mean? Should I have expected to get 100% across the board? This is from a TE410P running on Debian 2.6.11-1-686-smp on a dual Xeon 2.4GHz server. Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel.conf in /etc not /etc/asterisk - historical reason?
Thanks for getting back to me, the only reason that I see to move it (and more importantly to move it to /etc/asterisk) is that I am intending to use DRDB to make the machines as identical as possible, and to ensure that the configs of the two machines are kept in-sync. My mount points for the 3 replicated drives were going to be /etc/asterisk /var and /home (or /users) I cant replicate /etc as things need to be different in some of its child directories (init.d and sysconf are two) (although I guess I could as I'm not intending to replicate /var/spool/ and thats below var) If zaptel.conf moves to /etc/asterisk, it keeps my replication simpler than adding lots of mount points nb - DRDB is a replication technology (laymans term I know) (commonly used with linux-ha) I agree it doesn't belong in /etc/asterisk, but its convient, especially since I know of no other application that interfaces with it :) David On 5/16/05, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], David John Walsh [EMAIL PROTECTED] wrote: One thing gets me, and hopefully someone here can confirm my suspision - why is zaptel.conf not with the other asterisk files (I assume it is because its responsable for bringing up the hardware, not strictly part of the asterisk application) Yes. Zaptel came before Asterisk and is independent of it. It is possible for other non-Asterisk software to make use of Zaptel, without Asterisk needing to be present at all. Would someone care to confirm my suspision, and if I'm wrong advise me why. As a follow on to this - if i were to move it somewhere else, is it the somthing.c file that would need to be changed to reflect this move. Don't know, but I have trouble understanding the need to move it. The only place that it would make sense to move it to would be /etc/zaptel/zaptel.conf, but since it is a single file, why bother? It certainly doesn't belong in the /etc/asterisk directory. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] POE hub
I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Hope you really meant a cheap switch... you don't want to use hubs of any sort in the asterisk environment since hubs are limited to half-duplex ethernet connections. Check the NetGear stuff. I believe someone indicated they are selling a poe switch for something like $400. Can you actually buy hubs these days? With an 8 port switch costing $25, there doesn't seem much of a niche opportunity for a cheaper option. Yes, one can still buy new hubs. We typically use them in conjunction with Sniffer or Ethereal when clients don't have a switch with port mirroring cabilities. But, I'd suspect the OP really meant a switch. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home 1.0 + Sipgate UK/SIP Provider
-- Executing Dial(SIP/201-fcb3, SIP/sipgate/###) in new stack -- Called sipgate/## Paul I apreciate why you've the dialled digits out there, but would you be good enough to include the first few, as if your asterisk box is sending extra / unwanted / too few digits to sipgate its never going to work :) Other than that it seems someone else has posted config for your reference to check. David ___ 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] POE hub
Hope you really meant a cheap switch... Yes, I am going to use 16 port Linksys switches if I can't get POE units at a reasonable price. Looks like the 8-port Netgear FS108P is about $103 to $141 right now, and it supposedly supports poe. ___ 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] zttest
On May 16, 2005, at 14:37, Rich Adamson wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? If this is a Linux box, look at the kernel boot arguments in [lilo| grub].conf and append noht, that disables it. My grub.conf on one of my boxes looks like this: title CentOS (2.4.21-27.0.4.ELsmp) root (hd0,0) kernel /vmlinuz-2.4.21-27.0.4.ELsmp ro root=LABEL=/ noht initrd /initrd-2.4.21-27.0.4.ELsmp.img jens ___ 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] POE hub
D-Link makes a whole line of them. http://www.provantage.com/YDLNS046.htm On 5/15/05, Chris Mason [EMAIL PROTECTED] wrote: I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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] Broadvoice: No Service, No Email reply but charging the credit card still works
I cannot email them, I cannot call them, I do not get an answer, but the credit card is still charged, although NO phone calls are possible anymore, ... Are they still in business? (except charging credit cards) bye Ronald ___ 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] POE hub
The 8 port would only be 7 port after uplink, so even two of them is not going to give me 16 ports, so they are not suitable, I don't have room for three devices. Shame. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, May 16, 2005 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] POE hub Hope you really meant a cheap switch... Yes, I am going to use 16 port Linksys switches if I can't get POE units at a reasonable price. Looks like the 8-port Netgear FS108P is about $103 to $141 right now, and it supposedly supports poe. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: zaptel.conf in /etc not /etc/asterisk - historical reason?
On 13:24, Mon 16 May 05, David John Walsh wrote: Thanks for getting back to me, the only reason that I see to move it (and more importantly to move it to /etc/asterisk) is that I am intending to use DRDB to make the machines as identical as possible, and to ensure that the configs of the two machines are kept in-sync. My mount points for the 3 replicated drives were going to be /etc/asterisk /var and /home (or /users) I cant replicate /etc as things need to be different in some of its child directories (init.d and sysconf are two) (although I guess I could as I'm not intending to replicate /var/spool/ and thats below var) If zaptel.conf moves to /etc/asterisk, it keeps my replication simpler than adding lots of mount points nb - DRDB is a replication technology (laymans term I know) (commonly used with linux-ha) I agree it doesn't belong in /etc/asterisk, but its convient, especially since I know of no other application that interfaces with it :) Move it to /etc/zaptel/zaptel.conf and make a symlink to it in /etc. That way you can sync /etc/zaptel and also be sure no application fails because of the missing file. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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] POE hub
Moreover, The FS108P can only power 4 ports simultaneously. I'd prefer something like this: http://www.netgear.com/products/details/FSM7326P.php Or a Cisco equivalent. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason (Lists) Sent: 16 May 2005 13:48 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] POE hub The 8 port would only be 7 port after uplink, so even two of them is not going to give me 16 ports, so they are not suitable, I don't have room for three devices. Shame. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, May 16, 2005 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] POE hub Hope you really meant a cheap switch... Yes, I am going to use 16 port Linksys switches if I can't get POE units at a reasonable price. Looks like the 8-port Netgear FS108P is about $103 to $141 right now, and it supposedly supports poe. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: chan_capi, chan_misdn and chan_modem
For example, if you use an Point-to-Multipoint ISDN connection (not 'Anlagenanschluss'), then you won't get an immediate 'BUSY' on SIP Busy/Congestion. It's not possible to signal the caller 'Busy' or 'Reject', because there is a timeout on the ISDN-Bus for ANY OTHER device which may answer the call. Only on timeout, the Busy is signaled. So what type of connection and environment do you use? I'm using - p2p (Anlagenanschluß) - Eicon Diva-Server 4BRI - Linux 2.6.11 Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
How about this... Replace the old text in /usr/src/zaptel/zaptel.conf.sample: # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 with this text: # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # All T1/E1 spans generate a clocking signal on the transmit side. The # timing parameter determines whether the clocking signal from the opposite # end of the T1/E1 is used to sync our clock. For T1/E1's connected to a # pstn provider (telco), chose 1 for using this T1 as the primary clock, # 2 for a secondary (if multiple T1/E1's are in use and the second T1 is # to be used for clock sync should the primary fail), or 3 for the next # T1, etc. If the T1/E1 is connected to a channel bank or if the T1/E1 # is not to be used for clock sync, then specify the timing as 0. (A quad # T1/E1 card should only have a single T1/E1 specified as timing = 1, or # primary clock sync. Incorrect timing sync may cause clicks/noise in # the audio, poor quality faxes, or fax failures, etc.) If this is helpful, I'll submit a bug for the text. Thoughts anyone? Very good. Now I'll try to muddy the waters with my own ignorance on this subject. Where is the clock source that the T1/E1 board, with 0 for timing, uses to generate the tx data stream? Is there a PLL on each board? Or is some central source used? For example, I have one system with two separate T100P cards--one for a telco T-1 (#1) and the other for a channel bank (#2). For timing, #1 (telco) is set to 1 and #2 (channel bank) is set to 0. How does card #2 get its timing to generate its tx stream? Does card #1 interrupt the CPU based on the retrieved clock stream, and the CPU drive the other boards based on #1's interrupts? ___ 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] 64 bit
On 5/13/05, Kaj J. Niemi [EMAIL PROTECTED] wrote: How did you get it to compile? Do you have to have a strictly 64 bit compile environment? On RHEL4 it compiles just fine out of the box. Some of the locations are not strictly correct (things get sent to /usr/lib instead of /usr/lib64..) but those are easily fixed when building the rpms. Everything is strictly 64-bit, running mixed 32/64 is just asking for trouble. I also integrated the building of pwlib 1.9.0/openh323 1.17.1 to the whole build process and sound between sip - h.323 users behing Cisco CME systems works great along with using res_config_mysql, cdr_addon_mysql for realtime and cdr logging. :) I spent a few days to figure out the best way of building everything and now usually just drop a cvs snapshot (and/or selective patches from mantis) to stay current. // kaj ___ I am running on SUSE 9.3x64 with very good results. zttest shows 99.9877 (lowest) However the distro supplied the 64bit rpms A.G. (Tony) Nichols I.S. Manager ___ 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] callback problem
This is a portion of code out of a callback program I'm using:
if ($response eq 1) {
verbose(CALLBACK: Callback to $clidnumber confirmed.);
$out = new Asterisk::Outgoing;
$out-setvariable(Channel, $channel . $clidnumber);
$out-setvariable(MaxRetries, 1);
$out-setvariable(context, $context);
$out-setvariable(extension, $extension);
$out-setvariable(CallerID, $outgoingclid $clidnumber);
$out-outtime(time() + 15);
$out-create_outgoing;
$AGI-stream_file($dir/callback-confirmed);
} else {
verbose(CALLBACK: Callback to $clidnumber canceled!);
$AGI-stream_file($dir/canceled);
};
Note the outtime line. Change the 15 to however many seconds you want
to wait. If you would like the entire file, let me know.
Darren Wiebe
[EMAIL PROTECTED]
Kamran Ahmad wrote:
hello
i am trying to make a callback solution.
client will call callback number and call is
terminated.
now callback server will create a call for that
client.
actually i have a problem in this process. that server
is creating call to client (UA) when previous call is
not disconnected yet.
UA--Asterisk(callbacknumber) callis answered
UA--Asterisk(callbackserver) call is created
when previous call is not hangup
any solution how to resolve this problem
i am stuk here
can any one help me in this
Kamran
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Re: [Asterisk-Users] cisco 3620 setup (newbie cisco alert)
Your configuration is OK. Cisco is counting from 0, so Serial 0:15 is 16th channel (D-channel) of first E1 (if you don`t have serial interfaces also...). zaptel/asterisk is counting from 1, so 1-16 is D-channel of first E1 interface. See archive for thread named Asterisk and Cisco AS5300 or 3600. There is example of configuration. -b - Original Message - From: Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 16, 2005 2:18 PM Subject: [Asterisk-Users] cisco 3620 setup (newbie cisco alert) I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots D-Channel Configurable modes Status E1 0/0 3115pri/channelized Administratively up E1 1/0 3115pri/channelized Administratively up I have set both controllers to hdb3, ccs with crc4 I notice that the D-Channel is set to 15. However, in the zaptel.conf file I use with my TE410p card, the D-Channel is set to 16. So, is cisco counting 0-14 (hence 15 is the D-Channek) and zaptel counting from 1-15 (and therefore 16 is the D-channel) ? Or is my config wrong on the cisco or zaptel ?? Any help would be welcome. Thanks ! Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_config_mysql.so relocation error
list wrote: Hi, in my attempt to install ISDN BRI card, I loaded asterisk-addons. I think I went to fast and buggerd up the locations of the files and directories. cant load asterisk again, getting: [res_config_mysql.so] = (MySQL RealTime Configuration Driver) asterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_config_load Ouch ... error while writing audio data: : Broken pipe To revert; it should be ok for me to do cvs update on zaptel libri asterisk only? Or how can I repair my cause of relocation error? You need to make clean everything and re-make. Are you using CVS-HEAD for everything? res_config_mysql is NOT for 1.0.7. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On Mon, 16 May 2005, Michael Welter wrote: Where is the clock source that the T1/E1 board, with 0 for timing, uses to generate the tx data stream? Is there a PLL on each board? Or is some central source used? For example, I have one system with two separate T100P cards--one for a telco T-1 (#1) and the other for a channel bank (#2). For timing, #1 (telco) is set to 1 and #2 (channel bank) is set to 0. How does card #2 get its timing to generate its tx stream? Does card #1 interrupt the CPU based on the retrieved clock stream, and the CPU drive the other boards based on #1's interrupts? The #1 card will derive its clock from the received stream from the telco. The #2 card will run on an internal free running clock. The two cards are not synchronized at all. For the 4-port cards there is an unused connector marked timing. Perhaps Digium intends to update the fpga to allow cross-board timing distribution at a later date. It is possible, though complicated, to synchronize the 2Mbit clocks on two unrelated cards by measuring the accumulated phase shift (difference in interrupt rate) over time and compensating, thus implementing a PLL in software. Digium has not shown any intereset in such a solution. It is not clear if the internal hardware clock generator can be fine tuned enough to implement this. Peter ___ 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] 2 servers via PRI
Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to pri_net...this cant be all? And the cable pin1 -- pin4 pin2 -- pin5 pin3 -- pin6 pin4 -- pin1 pin5 -- pin2 pin6 -- pin3 pin5 -- pin8 pin8 -- pin7 Please Help and advice Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
Peter Svensson wrote: On Mon, 16 May 2005, Michael Welter wrote: Where is the clock source that the T1/E1 board, with 0 for timing, uses to generate the tx data stream? Is there a PLL on each board? Or is some central source used? For example, I have one system with two separate T100P cards--one for a telco T-1 (#1) and the other for a channel bank (#2). For timing, #1 (telco) is set to 1 and #2 (channel bank) is set to 0. How does card #2 get its timing to generate its tx stream? Does card #1 interrupt the CPU based on the retrieved clock stream, and the CPU drive the other boards based on #1's interrupts? The #1 card will derive its clock from the received stream from the telco. The #2 card will run on an internal free running clock. The two cards are not synchronized at all. For the 4-port cards there is an unused connector marked timing. Perhaps Digium intends to update the fpga to allow cross-board timing distribution at a later date. It is possible, though complicated, to synchronize the 2Mbit clocks on two unrelated cards by measuring the accumulated phase shift (difference in interrupt rate) over time and compensating, thus implementing a PLL in software. Digium has not shown any intereset in such a solution. It is not clear if the internal hardware clock generator can be fine tuned enough to implement this. How can that work? You can measure the error, but you have no ability to tweak the clock from software. Two cards could only be synced by hardware. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: callback problem
hello he is still not replying after correct time this is the sip debug May 16 21:41:02 WARNING[3902]: chan_sip.c:730 retrans_pkt: Maximum retries exceeded on call 76fa142e2805cc9a5d44ba4564165b1e@ for seqno 102 (Critical Request) May 16 21:41:02 NOTICE[3902]: pbx_spool.c:234 attempt_thread: Call failed to go through, reason 1debug May 16 21:41:03 NOTICE[3902]: sched.c:290 ast_sched_del: Attempted to delete non-existant schedule entry 15! __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ 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] Broadvoice: No Service, No Email reply but charging the credit card still works
Broadvoice turned out to be a very frustrating and disappointng service. I gave up on them a few weeks ago and cancelled my account. I signed up with VoicePulse, but the olny e-mail I received from them stated there were problems signing me up and to call a number. I have not and probably will not follow through and they have not sent me any more e-mails... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Monday, May 16, 2005 5:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Broadvoice: No Service, No Email reply but charging the credit card still works I cannot email them, I cannot call them, I do not get an answer, but the credit card is still charged, although NO phone calls are possible anymore, ... Are they still in business? (except charging credit cards) bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: failure notice
Can we get this looser bumped, this has been happening for the last 2 weeks now. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:MAILER- [EMAIL PROTECTED] Sent: Monday, 16 May 2005 12:02 AM To: Dean Collins Subject: failure notice Hi. This is the qmail-send program at smtp.register.it. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out. [EMAIL PROTECTED]: This message is looping: it already has my Delivered-To line. (#5.4.6) --- Below this line is a copy of the message. ___ 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] POE hub
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Giles Coochey Sent: Monday, 16 May 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] POE hub Moreover, The FS108P can only power 4 ports simultaneously. I'd prefer something like this: http://www.netgear.com/products/details/FSM7326P.php Or a Cisco equivalent. [DC] Lol - yeh and at $1300 I prefer some power plugs. Dean ___ 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] Need off-the-shelve PC for Asterisk Server
Does any one have any recommendations on an off-the-shelve PC for an Asterisk Server? This is for a proof of concept, so it needs to be inexpensive. I have tried 2 different PC's and had problems with the sound cards. I am thinking of PC's I can buy from local dealers like Best Buy, Office Depot. SO a cheap HP, Compaq or eMachine would work fine for me. --Steve ___ 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] .call file
hello can any one tell me Channel: SIP/[EMAIL PROTECTED]:5060 MaxRetries: 1 # Retry in 5 min RetryTime: 60 WaitTime: 30 Context: default Extension: 6000 Priority: 1 why this is not working Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html ___ 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] FW: failure notice
Dean Collins wrote: Can we get this looser bumped, this has been happening for the last 2 weeks now. I hate this kind of thing as much as anyone, but isn't bumping him off a bit extreme? :-) Regards, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:MAILER- [EMAIL PROTECTED] Sent: Monday, 16 May 2005 12:02 AM To: Dean Collins Subject: failure notice Hi. This is the qmail-send program at smtp.register.it. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out. [EMAIL PROTECTED]: This message is looping: it already has my Delivered-To line. (#5.4.6) --- Below this line is a copy of the message. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
How about this... Replace the old text in /usr/src/zaptel/zaptel.conf.sample: # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 with this text: # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # All T1/E1 spans generate a clocking signal on the transmit side. The # timing parameter determines whether the clocking signal from the opposite # end of the T1/E1 is used to sync our clock. For T1/E1's connected to a # pstn provider (telco), chose 1 for using this T1 as the primary clock, # 2 for a secondary (if multiple T1/E1's are in use and the second T1 is # to be used for clock sync should the primary fail), or 3 for the next # T1, etc. If the T1/E1 is connected to a channel bank or if the T1/E1 # is not to be used for clock sync, then specify the timing as 0. (A quad # T1/E1 card should only have a single T1/E1 specified as timing = 1, or # primary clock sync. Incorrect timing sync may cause clicks/noise in # the audio, poor quality faxes, or fax failures, etc.) If this is helpful, I'll submit a bug for the text. Thoughts anyone? Very good. Now I'll try to muddy the waters with my own ignorance on this subject. Where is the clock source that the T1/E1 board, with 0 for timing, uses to generate the tx data stream? Is there a PLL on each board? Or is some central source used? Each T1/E1 card has a clock on board and most use a crystal in the circuity. However, the frequency of that clock is not always right on and may drift by a few cycles/second or more. Regardless of how the clock is implemented on the card, its important to sync that clock to the telco so that your T1/E1 transmit data is sent to the telco at the same rate as what they are sending data to you. (The telco engineers understand this very very well, and have been syncing their clocks to higher-level switching offices for years.) If you don't sync your on- card clock to theirs, then you will be transmitting pcm data to their site at some rate they aren't expecting which will cause some errors to occur depending upon how far off your clock happens to be. Of coarse, they are the only ones that will see those errors (since it is on their receive side), and the telcos won't call you to tell you that is actually happening. For the most part, they don't even monitor the incoming errors unless asked to. If your T1's don't connect to a telco (pstn line), then the clock sync is less of an issue. You just have to decide which end of the T1 is going to be the master clock, and make all attached devices subservant to it. (The subservant devices would then have timing=1, as an example.) For example, I have one system with two separate T100P cards--one for a telco T-1 (#1) and the other for a channel bank (#2). For timing, #1 (telco) is set to 1 and #2 (channel bank) is set to 0. How does card #2 get its timing to generate its tx stream? Does card #1 interrupt the CPU based on the retrieved clock stream, and the CPU drive the other boards based on #1's interrupts? It doesn't make any difference. The pcm data that arrives from the telco is buffered in the zaptel and/or asterisk code, and sent out the second T1 card as soon as it can. That buffering reduces (or eliminates) the need to sync one T1 card to another. However, if the clock on the second card were way off frequency, there could be a missed pcm frame from time to time. The missed frame would not even be noticed by users in most cases. (There was some discussion about how great of an impact that really has several months ago. Personal opinion plus experience says its a non-issue other then on some very extreme cases where a clock is way off frequency. That tends not to happen with today's electronics.) If you think about the variable delays that occur because of contention for the pci bus, interrupt latency, etc, there are likely larger swings in dropped packets resulting from that then there would be from multiple T1 cards not having the exact same clock frequency. The bigger issue with multiple T1's occurs with the Quad T1 cards, where a single on-card clock is used with all four T1's. Syncing that clock to the pstn is the only correct way (timing=1), making all other downstream T1's (eg, channel banks, pbx's, other * boxes) subservant to it (timing=0). If you had multiple asterisk boxes located in several cities, and each of those asterisk boxes had a pstn T1, then you better read and understand T1 clock syncing very well as it will make a signifcant difference how each box obtains clock sync
[Asterisk-Users] problems with asterisk starting from init.d
Hi All I had asterisk running on a xercom install, I upgraded the box to a full debian install and now asterisk is not starting from on boot. I can start asterisk from the command line fine no problems, but when i type /etc/init.d/asterisk start it says asterisk PBX started. It doesn't start it though, when I look at the log file it has this. May 16 10:19:05 WARNING[3711]: Unable to open '/dev/zap/channel': Permission denied May 16 10:19:05 ERROR[3711]: Unable to open channel 1: Permission denied here = 0, tmp-channel = 1, channel = 1 May 16 10:19:05 ERROR[3711]: Unable to register channel '1' May 16 10:19:05 WARNING[3711]: chan_zap.so: load_module failed, returning -1 May 16 10:19:05 WARNING[3711]: Loading module chan_zap.so failed! the card works fine when I start asterisk from the command line. Can anyone help me please? Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.10 - Release Date: 5/13/2005 ___ 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] AreskiCC
My 2 cents: If I dont misunderstand it, I would guess youll have to read it again and type in myroot AND mypassword or what it says and NOT your_login and your_password. After that, you can change it to your preferred password etc... Thats what I guess...But Im not sure. Regards //Stefan On 5/16/05, Robson Ribeiro [EMAIL PROTECTED] wrote: Hi, I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and Postgres 7.4. First of all the instructions are very confusing and hard to follow if you are not an expert. But, I managed to install it andobviously t doesn't work. The other instructions I found on wiki are a great effort but incomplete. Basically the first thing that happens is that when I load /areskicc/Public/index.php it refuses my username and passwork (AUTHENTICATION REFUSED, please check your login/password! ) which I guess is the same as the one I configured on defines.php right?) and after I reinsert it I get the error: Method Not Allowed. The requested method POST is not allowed for the URL /areskicc/Public/index2.php. In any case, does anybody know of any better instructions on how to install and configure AreskiCC? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1-800 with FWD
oh, Thank you !! Problem solved. Juanjo On 5/13/05, Patrick M. Gray, Jr. [EMAIL PROTECTED] wrote: Did you dial the 800 number correctly? You need to dial *1800XXX. I had this problem for a while and then checked out the docs on FWD's website. Any toll-free number seems to require a * before dialing. You can setup your dialing prefixes to add it automatically so it becomes transparent to users. Pat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juanjo Portela Sent: Friday, 13 May, 2005 19:07 To: Lista Asterisk Subject: [Asterisk-Users] 1-800 with FWD Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial(SIP/Phone4-e85b, SIP/[EMAIL PROTECTED]|90|Ttr) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 500 I'm terribly sorry, server error occured (1/SL) back from 69.90.155.70 -- SIP/fwd.pulver.com-f526 is circuit-busy == Everyone is busy/congested at this time Have you any idea? Thank you in advance, Juanjo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Satellite Providers
I am operations vp for a wholesale VOIP network and we have customers sending us VOIP over satellite that works quite well.Several well known carriers just do not work for VOIP in my experience. [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Wednesday, May 11, 2005 19:11 To: Chad Wicker Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Satellite Providers I don't doubt at all what you are saying. We never tested a truly high-end solution such as the one you described, because the cost would have been prohibitive for our application. I'm sure we only evaluated shared solutions. I guess my mistake was believing the CIR claims. At the really low-end, I didn't expect much, since they don't offer ANY CIR. But when they claimed 64k, silly me, I believed it. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 11 May 2005, Chad Wicker wrote: Well there are several problems in your description of Satellite services. For one you are grouping several differing technilogies together as one. What it seemed like you were testing was a shared bandwidth solution typically used by providers to reduce cost. It isn't uncommon to experience sever delays and packet loss on these types of systems. Alot of these shared providers claim 64k cir then oversubscribe over that. Lies, yes, theift yes, and they get away with it... What you would want to ask for is a SCPC (Single Carrier Per Channel) circuit and you should have much better results, cost? a lot more than these shared solutions. You may want to look into the maritime providers/teleports in the area for this type of service. Delay for a decent circuit should not be over 600 ms and it should be steady. Proof is in the pudding, in a SCPC circuit with a v.35 interface you can run an extended BERT test on it without error. and that's Sync data... I speak confidently on this as we are a provider of VSAT services in the oilfield industry. We are bombarded with these low cost competition and have to defend ourselves daily. Alot of providers sell crap at a decent price. We don't and won't. It hurts our market penetration but we tend to keep customers for a good long time. I can answer a lot of questions on this subject if anyone needs. It's a lot like point to point microwave, they experienced their bandwidth sharing days and they quickly died on the vine. The driving force behind shared solutions is that satellite bandwidth is expensive. Chad C. Wicker Systems Engineer Petrocom [EMAIL PROTECTED] 5/11/2005 1:06:52 PM We looked at this earlier this year and, after evaluating several companies, could not get it to work well enough. The problem didn't seem to be latency, but rather lost packets in the upstream direction. Most of the time, we couldn't even get the phone to register, but even when we could, there was such a large amount of breakup (in the up direction) that it was nearly unusable. We tried low-end, consumer type services and they didn't work at all. Even the high-end services that claim to offer guaranteed bandwidth apparently do not live up to their claims. We tried running G.729, which should only need about 32-40k over a link that claimed to guarantee 64k, and the best we got was broken sound. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 11 May 2005, Yiannis Costopoulos wrote: Hi All, I am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis. ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C200 5-05-11%5Cc819e577de1140fbaa62d0a53c83de86C=2 -- -- - This message has been inspected by DynaComm i:mail -- - ___ 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] Giving user progress in an voice menu system
Hi Josiah
Thanks for the info. What I decided to do instead was to modify my own
macro so I could pass the ring type to it. It may have helped me had I
remembered that the default config comes with a dial macro, but then
probably not, as I rewrite things all the time.
I like to reinvent the wheel.
Anyway, I didn't think much about it at the time, because the problem
got solved, but I'll post my macro here in case it's useful to anybody
else. Keep in mind it's fairly limited in scope, and it depends on
outside variables that are common throughout my extensions.conf file. I
can't see it being terribly useful for anybody here, but what do I know?
[macro-ext]
;; ${ARG1} is the sip channel to dial, ${ARG2} is the dial type.
;;Most times, that's simply a ring. For the menu system, I have it play
music on hold while it tries an extension.
exten = s,1,Dial(${IN_CHAN}/${ARG1}|${IN_TO}|${IN_OPT}${ARG2})
exten = s,2,VoiceMail(su${ARG1})
exten = s,3,Hangup
exten = s,102,VoiceMail(sb${ARG1})
exten = s,103,Hangup
Josiah Bryan wrote:
On Thursday 12 May 2005 3:43 pm, Sean Kennedy wrote:
Hi all,
I have a voice menu system ( Outlined below ), and I'd like to give the
user some feedback when they dial an extension ( ringing, music,
SOMETHING ). As it stands, when a user enters an extension from the
menu system, they hear silence while the line rings. I even tried
including the Ringing application before calling my macro to dial the
phones, with no luck.
Any help is apprecaited.
Odd - my receptionist was having a similar problem. I used the stdexten macro
that came with the demo files - when ever someone dialed directly (inside) or
directly thru the IVR (no receptionist pickup) - the ringback was fine. But
when the receptionist picked up and transfered - no ringback. All three
methods of dialing went thru the stdexten macro - very puzzling. The solution
I finally came up with was to add the 'm' option to the 'Dial' command.
Code speaks louder than words, so here you go..its obviously modified a bit -
but all should be self explanitory. The SIP/op channel is our receptionist
phone. The macro only adds the MOH option if the call is from the
receptionist phone, otherwise it leaves all options at default.
Anybody else have any other solutions or need debug outputs to figure this
out?
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Re: [Asterisk-Users] SIP Gerenal settings conufsion
Jeffrey Starin wrote: Jonathan! You don't know how much that simple explanation has helped me understand Asterisk. Well done. Well said. And to the point clearly. I would hope this could find it's way onto the Asterisk Wiki and be the *first* thing someone reads when looking at the documentation about sip. Thanks a million! blush There are many things I've found in Asterisk so far that take a while to wrap one's brain around. Once the effort is made, though, it's definitely worth it. Hopefully one day the learning curve won't be quite as steep. You'll find that once things start falling together in your mind, it gets a lot easier. And really fun, too, if you're into that sort of thing. Reminds me of early-90s Linux--and look where that is now. -Johnathan ___ 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] zttest
That's a setting of the BIOS (at least on the motherboard we have). - Waldo On May 16, 2005, at 8:37 AM, Rich Adamson wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). How do you disable hyper threading (what's the command and where is it placed)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need off-the-shelve PC for Asterisk Server
Dell's entry level line of servers is very Linux friendly. I use poweredge for some production systems (yes, even with a single drive) but if is only for a proof of concept, then a $50 Compaq deskpro which are also Linux friendly might be an option. Stephen McAllister wrote: Does any one have any recommendations on an off-the-shelve PC for an Asterisk Server? This is for a proof of concept, so it needs to be inexpensive. ___ 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] Broadvoice: No Service, No Email reply but charging the credit card still works
Ronald Wiplinger wrote: I cannot email them, I cannot call them, I do not get an answer, but the credit card is still charged, although NO phone calls are possible anymore, ... Hmm. I called them twice yesterday to ask questions, the queue wait was less than a minute in both cases. First time was via their own service, second time was via my cell phone. Their techs were very friendly and accommodating (but not the most knowledgeable.) My service is working normally. In fact, now that things have settled out with their new partners, it seems to be working *better* than before--faster call completions, better voice quality. Network probes show 1% packet loss to their lax and dca proxies and back over a 48 hour period. They emailed me some follow up information after my last call; it arrived a few minutes later. So yes, they are still in business. (They were within moments of losing *my* business when all the lights came back on last week. Guess I'm just a sucker for new technology and unlimited free phone calls :-) Can you elaborate on what is happening with you? -Johnathan ___ 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] zttest
This is interesting. Do you also have a TE410P? - Waldo On May 16, 2005, at 2:46 AM, Wilson Pickett wrote: After I run it, I get the following: 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% Just for reference, I'm running a PIII-800Mhz and I get (with no particular load on CPU) -Best: 100.00 -- Worst: 99.987793 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 99.987793% 99.987793% 99.987793% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: failure notice
On Mon, 2005-05-16 at 22:21 +0800, Steve Underwood wrote: Dean Collins wrote: Can we get this looser bumped, this has been happening for the last 2 weeks now. I hate this kind of thing as much as anyone, but isn't bumping him off a bit extreme? :-) The account doesnt exist, he cant confirm a delete from the list, may have forgotten anything since he doesnt get email there anymore ... -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] zttest
Thanks. That gives me something to work on. - Waldo On May 16, 2005, at 4:59 AM, Damian Funnell wrote: Hi Waldo, I would be money on your problem being related to the accuracy of zttest. One way of checking IRQ's is to run cat /proc/ interrupts, but it is a lot more accurate to run lspci -v and lspci -vb. I would recommend Googling the lspci command, although the output is pretty self explanatory. The TDM appears as a TigerJet card, not sure what TE410P will list as. PCI devices have their IRQ's dictated by the BIOS of the host system. How (and if) you can configure these manually depends on the type of BIOS you have... in our IBM xSeries 206 we had to actually juggle cards between slots to get it to assign a unique IRQ to the TDM400P. Good luck! D. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Waldo Rubinstein wrote: Damian, Thanks for your input. Hyperthreading is in fact enabled and now that you mention this I will disable it. The reason I ask is because under some load (may be 40 simultaneous calls), voice quality degrades. We have audio problems where one party hears the other but not viceversa and then it all works fine. It's random audio quality problems in general. During these cases, I'm constantly running vmstat 1 and CPU utilization is always 85%+ idle. I will also look into setting the TE410P in its own IRQ. Do you know how I can do that? Is that a motherboard BIOS setting or is it something that needs to be done to the TE410P itself? Thanks, Waldo On May 16, 2005, at 12:59 AM, Damian Funnell wrote: Hi Waldo, it really depends on who you ask - Digium say that anything less than 99.99% is going to result in problems, but ours regularly runs at around 99.98% and we don't have any problems. One of our boxes was running at around 99.96% and we had major issues with the voice quality packing up from time to time. We disabled hyper threading and put the TDM400P on its own IRQ and the results came back up over 99.98% (haven't had any problems since). Do you have issues with your * box? If so then I would start worrying about zttest output (and thinking about disabling hyper threading on those dual Xeons), otherwise have a smile and a beer and pity us poor fools who have had problems due to poor results. Cheers, Damian. FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Waldo Rubinstein wrote: I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% --- Results after 57 passes --- Best: 100.00 -- Worst: 99.975586 -- Average: 99.987793 What does this mean? Should I have expected to get 100% across the board? This is from a TE410P running on Debian 2.6.11-1-686-smp on a dual Xeon 2.4GHz server. Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] xbox asterisk?
http://www.pbs.org/cringely/pulpit/pulpit20050512.html interesting comment this week about the Xbox any intelligent thoughts here? I know the price point puts it above most users Asterisk outlay (I run mine on a $100 P3 -800) But interesting to see what happens if people start running video conferencing etc on their home asterisk servers, and lets face it where else can you buy this kind of subsidized processing power from at that price. Cheers, Dean ___ 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] FXO/FXS suggestions:
Rich Adamson wrote: Im looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. Your only choice for zaptel type is the TDM card. Probably the next best choice is the spa3000. I wish I could find out for sure how well the spa-3000 FXO works with * and same for the Grandstream Handytone 488. I need FXO-SIP conversion in places where I don't need a PC. The spa3k works pretty good in most cases; been using one for several months without any major issues and know of lots of others doing the same. It can be a little time consuming figuring out the many config options available though. (Use the tool at voxilla.com to help, and then review what the tool changed as sort of an educational thing.) Its my understanding from others the 488 does not support the fxo port as an addressable sip port from *. Its only accessible from the fxs port, apparently as a fallback or something like that. I've got my spa3k set up sort of like a mini pbx. The dialplan within it is configured to send all calls starting with an 8 out the voip port, while all other calls default to fxs - fxo dialing. No remedial training required for non-technical users, and * is not in the middle of normal local calls. Another spa3k is configured for the fxs port to register with *, and the fxo port also registers with asterisk, forcing asterisk to be in the middle of all calls. In this case, one must use the g711 codec as apparently there isn't enough horsepower to run to simultanous g729 sessions at the same time. For this config to function, one has to essentially call-forward any incoming fxo calls to asterisk. Not a big deal at all; works fine. Given the cost of the spa, try one to see if it meets all of your requirements. I really appreciate your reply and those of others who have shared their experiences with the SPA-3000. I have had good experiences with my other SPA- devices so I will be ordering an SPA-3000. Maybe Sipura will cook up some multi-FXO products in the future. Meanwhile, I would consider a stack of 4 SPA-3000 units to be a reasonable cost solution even if I didn't need the 4 FXS ports. I actually plan to always config 1 FXS port at any location for techs that are there. I get bad results with cell phones in server rooms so the FXS port is a bonus. ___ 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] Vonage users with Asterisk in UK?
Hi, I'd be interested in comments from any users of the vonage service in the UK? http://www.vonage.co.uk is the website. Where are the servers located, traceroute would be useful. What is the general reliability like? Thanks Mike ___ 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] FXO/FXS suggestions:
On Sun, 15 May 2005 23:37:28 -0500, Jon Gabrielson wrote: On Sunday 15 May 2005 09:53 pm, Paul wrote: Do you have the clout to get a handytone for evaluation and not have salespeople calling you every day to ask how it's going? :) Why not just buy one? You can buy one for less than $100 and if you don't like it, you can just turn around and sell it on ebay and get most if not all of your money back. What are the chances of someone who can't spend $100 on a item making it worthwhile for a company to spend time and money giving them an evaluation unit. I had an SPA-300...and just sold it off on Ebay. I replaced it with a TDM400 cards, which I just took out of service as well. The big issue with the SPA-3000 for me was the combination of echo and analog line level on the FXO. I could never get it quite loud enough and was always straining to head while jacking up the volume on my Polycom IP 600 phones. My solution was a hack that was supposed to be short term, but has worked so well I may stick with it. I had call forwarding put on my one remaining POTS line and forwarded it to a DID that I had setup through an ITSP but was not using heavily. Now all incomming analog calls go through that ITSP and actually arrive via my DSL. I can shut off call forwarding and answer the line with a backup analog handset if my DSL goes out. I had set this up to help me bridge a new server into production. That machine is a small form factor box running Astlinux. At the time Astlinux did not support the TDM card. It does as of the 0.2.6 release, but the call forwarding trick sounds better than the TDM. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] FW: failure notice
Steve Underwood wrote: I hate this kind of thing as much as anyone, but isn't bumping him off a bit extreme? :-) Hell no. Its a permanent error. It won't go away. Plus, this wastes digium's server time having to send back all the bounces. Bounce him off the list. Most mailing list servers have this feature built in where if a subscribed address bounces then it is blocked or removed automatically. -Matthew -Original Message- From: [EMAIL PROTECTED] [mailto:MAILER- [EMAIL PROTECTED] Sent: Monday, 16 May 2005 12:02 AM To: Dean Collins Subject: failure notice Hi. This is the qmail-send program at smtp.register.it. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out. [EMAIL PROTECTED]: This message is looping: it already has my Delivered-To line. (#5.4.6) --- Below this line is a copy of the message. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On Mon, 16 May 2005, Rich Adamson wrote: It doesn't make any difference. The pcm data that arrives from the telco is buffered in the zaptel and/or asterisk code, and sent out the second T1 card as soon as it can. That buffering reduces (or eliminates) the need to sync one T1 card to another. However, if the clock on the second card were way off frequency, there could be a missed pcm frame from time to time. The missed frame would not even be noticed by users in Any frequency error will eventually lead to dropped frames, and this is as it should be. For digital transfers this can cause problems. For voice it is normally a non-issue. Peter ___ 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] POE hub
Lol - yeh and at $1300 I prefer some power plugs. That's how I feel Chris Mason US Number: (646)722-0001 US Fax (815)301-9759 Skype: netconcepts ___ 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] cisco 3620 setup (newbie cisco alert)
Thanks for that - I've managed to configure the cisco box following various examples on the web, but come stuck at the following: dial-peer voice 100 pots application session max-conn 30 destination-pattern 0. translate-outgoing called 1 no digit-strip direct-inward-dial port 0:D forward-digits all ! the port command does not exist % Unrecognized command. Show Version gives #show version Cisco Internetwork Operating System Software IOS (tm) 3600 Software (C3620-J1S3-M), Version 12.3(13a), RELEASE SOFTWARE (fc2) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2005 by cisco Systems, Inc. Compiled Tue 26-Apr-05 09:12 by ssearch Image text-base: 0x60008B00, data-base: 0x619C ROM: System Bootstrap, Version 11.1(20)AA2, EARLY DEPLOYMENT RELEASE SOFTWARE (fc1) ROM: 3600 Software (C3620-J1S3-M), Version 12.3(13a), RELEASE SOFTWARE (fc2) delta uptime is 1 hour, 33 minutes System returned to ROM by reload System image file is flash:c3620-j1s3-mz.123-13a.bin cisco 3620 (R4700) processor (revision 0x81) with 58368K/7168K bytes of memory. Processor board ID 24364331 R4700 CPU at 80MHz, Implementation 33, Rev 1.0 Channelized E1, Version 1.0. Bridging software. X.25 software, Version 3.0.0. TN3270 Emulation software. Primary Rate ISDN software, Version 1.1. 2 Ethernet/IEEE 802.3 interface(s) 32 Serial network interface(s) 2 Channelized E1/PRI port(s) DRAM configuration is 32 bits wide with parity disabled. 29K bytes of non-volatile configuration memory. 24576K bytes of processor board System flash (Read/Write) Configuration register is 0x2102 Any more help appreciated :) Julian. barney wrote: Your configuration is OK. Cisco is counting from 0, so Serial 0:15 is 16th channel (D-channel) of first E1 (if you don`t have serial interfaces also...). zaptel/asterisk is counting from 1, so 1-16 is D-channel of first E1 interface. See archive for thread named Asterisk and Cisco AS5300 or 3600. There is example of configuration. -b - Original Message - From: Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 16, 2005 2:18 PM Subject: [Asterisk-Users] cisco 3620 setup (newbie cisco alert) I'm experimenting (using for the first time) with using a cisco3620 to connect to the PSTN via a channelised E1 interface, with * handling all of the SIP calls. If anyone has any installation tips / help / documentation I would be most appreciative :) However, my first question is this: when I am in the setup, I see the following: Current interface summary Controller Timeslots D-Channel Configurable modes Status E1 0/0 3115pri/channelized Administratively up E1 1/0 3115pri/channelized Administratively up I have set both controllers to hdb3, ccs with crc4 I notice that the D-Channel is set to 15. However, in the zaptel.conf file I use with my TE410p card, the D-Channel is set to 16. So, is cisco counting 0-14 (hence 15 is the D-Channek) and zaptel counting from 1-15 (and therefore 16 is the D-channel) ? Or is my config wrong on the cisco or zaptel ?? Any help would be welcome. Thanks ! Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing spool file ignored
trixter http://www.0xdecafbad.com wrote: How do you create them? There is a race condition with asterisk and the spool where if you create the file or copy it into the queue directory asterisk tries to read and parse the file before you have finished writing it. A suggested method instead is to create it on the same partition then move it into the appropriate directory to prevent this from occuring. Create the file somwhere else. Set the mtime (I think) to sometime in the future. Move the file to /var/spool/asterisk/ourgoing. Change the mtime to the current time or some time in the past. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Need off-the-shelve PC for Asterisk Server
Stephen McAllister wrote: Does any one have any recommendations on an off-the-shelve PC for an Asterisk Server? This is for a proof of concept, so it needs to be inexpensive. I have tried 2 different PC's and had problems with the sound cards. I am thinking of PC's I can buy from local dealers like Best Buy, Office Depot. SO a cheap HP, Compaq or eMachine would work fine for me. Or you can rent a cheapo dedicated box and have a play with that by just doing VoIP... -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ 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] FW: failure notice
Calling him a loser is a bit extreme. Maybe they fired him but he got a job that pays twice as much. Anyway, bumping him is not extreme at all. IIRC - some lists are setup to automatically unsubscribe people after N days of delivery failures. We only see this individually when we post but the list server is probably getting this for every new post to the list. Steve Underwood wrote: Dean Collins wrote: Can we get this looser bumped, this has been happening for the last 2 weeks now. I hate this kind of thing as much as anyone, but isn't bumping him off a bit extreme? :-) Regards, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:MAILER- [EMAIL PROTECTED] Sent: Monday, 16 May 2005 12:02 AM To: Dean Collins Subject: failure notice Hi. This is the qmail-send program at smtp.register.it. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out. [EMAIL PROTECTED]: This message is looping: it already has my Delivered-To line. (#5.4.6) --- Below this line is a copy of the message. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Scalability of chan_oh323
Michael Manousos wrote: Alistair Cunningham wrote: I have a customer who wants to do large volumes of H.323 to H.323 hairpinning. We haven't tested this scenario for large volumes before; maybe someone on asterisk-users has. If they buy a top of the line PC, how many concurrent calls are we likely to get? Routing logic will be simple, the machine won't be doing anything else, and let's assume no transcoding for now. We're not looking for an exact figure at this point, just a rough estimate for cost / benefit of Asterisk versus a proprietary system. Currently, without transcoding, you can get maximum 100 simultaneous H.323 channels per box. With the next release of asterisk-oh323 this number will be raised to ~180 channels. After that, major optimizations at the OpenH323 RTP/jitter buffer code are required to push this number up. Michael. That's a shame; my customer probably needs 400 to 500 channels (200 to 250 calls). Does anyone have experience of GNU Gatekeeper in proxy mode? Any idea of what load it can handle? -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.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] POE hub
If you shop that netgear option, you can get it under $1000, plus its managed so you can do things like VLANs and QoS which could come in handy. Another upside is that the Netgear will autodetect Cicso PoE vs. IEEE PoE (espeically important to me because I have a mix of 7900 phones and IEEE compliant PoE devices) Its expensive, but if you fill all 24 ports, its only $41 / port (you can uplink on one of its two GigE ports). ~Adam On 16/05/05, Dean Collins [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Giles Coochey Sent: Monday, 16 May 2005 8:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] POE hub Moreover, The FS108P can only power 4 ports simultaneously. I'd prefer something like this: http://www.netgear.com/products/details/FSM7326P.php Or a Cisco equivalent. [DC] Lol - yeh and at $1300 I prefer some power plugs. Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicepulse problems?
Hi all, Is anybody else experiencing problems with voicepulse? Today and over the weekend? I've had problems with both gateways, but one usually works when the other doesn't. I'm trying to eliminate my network from the problem. Sean ___ 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] POE hub
Well, I dont know the model numberr of the 3com poe hub that I used but it worked just fine with the polycom ip phones. Thank you, Steve Maroney On Mon, 16 May 2005, Chris Hills wrote: Steve Maroney wrote: The cheapest I have found was a 3COM 24 Port for $799.00. Thank you, Steve Maroney Be warned, we are a 3Com house, and I ordered a 4400 PWR to test it would work with our Siemens hard phones. Lucky I did, because it turns out they are not compatible! It seems the 3Com POE switches will only power 3Com devices. Instead, I ordered a bunch of PowerDsine injectors which work fine, and power a much greater range of devices. Regards -- Chris Hills IT Services North East Worcestershire College ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage users with Asterisk in UK?
On Mon, May 16, 2005 at 03:51:28PM +0100, Mike Dent wrote: Hi, I'd be interested in comments from any users of the vonage service in the UK? http://www.vonage.co.uk is the website. Where are the servers located, traceroute would be useful. What is the general reliability like? No idea re servers, you get a box, you plug it in to your broadband conneciton (and do a bit of configuration) and it just works. I installed mine this morning and it just worked, pretty good call quality. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 Skype/In callto://stevekennedyuk / UK callto://+442088167166 US callto://+13106518226mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.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
