[asterisk-users] 407 Proxy Authentication Required
Hello, I'm receiving some traffic from a Softwitch to Asterisk When I'm hiding the CallerID in the softwitch, everything is all right. When I allow to send the callerid from softwitch to Asterisk (actually, I would like to have it) Asterisk rejects the call with a 407 Proxy Authentication SIP packet. I copy-paste the SIP Invitation: -- Session Initiation Protocol Request-Line: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Method: INVITE [Resent Packet: False] Message Header Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK-27003710ff15ff5fff41 Transport: UDP Sent-by Address: 2.2.2.2 Sent-by port: 5060 Branch: z9hG4bK-27003710ff15ff5fff41 From: sip:[EMAIL PROTECTED];user=phone;tag=27003710ff15ff5fff41 SIP from address: sip:[EMAIL PROTECTED] SIP tag: 27003710ff15ff5fff41 To: sip:[EMAIL PROTECTED]:5060;user=phone SIP to address: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Sequence Number: 1 Method: INVITE Contact: sip:[EMAIL PROTECTED];user=phone Contact Binding: sip:[EMAIL PROTECTED];user=phone URI: sip:[EMAIL PROTECTED];user=phone SIP contact address: sip:[EMAIL PROTECTED] Max-Forwards: 10 User-Agent: x Cisco-Guid: 406000640-2566207248-2147483669-3311398977 Content-Type: application/sdp Content-Length: 164 -- sip.conf section: [2.2.2.2] host=2.2.2.2 type=friend insecure=yes context=test canreinvite=no (and calls goes to test context) Which header is forcing Asterisk to ask for authentication, and if I hide the callerid it's not asking it? Thanks, -- Carles Pina i EstanyGPG id: 0x17756391 http://pinux.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click to Dial
Hello, On Jul/24/2008, Brent Davidson wrote: I have a question about click to dial. Each of my users is going to have a VOIP phone with an assigned extension. Is there a simple way to build a web-based speed-dial list that will allow them to put in their extension, click on the number they want to dial, and have asterisk ring their phone, then as soon as they pick up, start dialing the number from the speed-dial? I think that a good start could be: http://lexatel.com/en/22/Whitepapers (currently, there is only one Whitepaper about click to dial, but of course you would need to do some frontend... easy to do, if you need any help feel free to ask). We are preparing some other Whitepaper too... Sorry for the semi-Spam :-) -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] next priority from Dial in Asterisk 1.6
Hello, I'm testing Asterisk 1.6 (from SVN). In my dialplan I have: -- exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg) exten = _00X.,2,Verbose(After Dial) -- If IP denies the call I receive: == Using SIP RTP CoS mark 5 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/401113-08200990, SIP/[EMAIL PROTECTED],,tTwWg) in new stack == Using SIP RTP CoS mark 5 -- Called [EMAIL PROTECTED] -- Got SIP response 484 Address Incomplete back from 212.121.243.35 == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (usuarios, 004477, 1) exited INCOMPLETE on 'SIP/401113-08200990' Why is not executing the Verbose after the Dial? Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] next priority from Dial in Asterisk 1.6
Hi, On Jul/23/2008, Carles Pina i Estany wrote: I'm testing Asterisk 1.6 (from SVN). In my dialplan I have: -- exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg) exten = _00X.,2,Verbose(After Dial) -- Also this doesn't work either: exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg) exten = _00X.,n,Verbose(After Dial) I mean, like before, after some SIP responses like 484 is not executing the after dialing command. In Asterisk 1.4.21.1 it was working as I expected. Is it a feature in Asterisk 1.6? or a bug? After 404 it's going to next priority, but not after 484. Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] next priority from Dial in Asterisk 1.6
Hi,
On Jul/23/2008, Anthony Francis wrote:
It is reading [EMAIL PROTECTED],,tTwWg as the device string.if you are
dialing to a sip connection called ip you would say
Dial(SIP/IP/${EXTEN},opts)
when I said IP i meant the IP value :-) not the two chars string IP.
Sorry for the confusion.
I would shoot my foot not! On Asterisk 1.6, if Dial fails the dialplan
goes to the next one (or n+101, etc.)
In Asterisk 1.6 it tries to go to the invalid extension:
[Jul 23 19:16:54] WARNING[10178]: pbx.c:3794 __ast_pbx_run: Channel
'Console/dsp' sent into invalid extension '555' in context 'usuarios',
but no invalid handler
(!!!)
I'm very sure that the same case (doing the Dial, but Dial is not
working) it goes to n+1.
Sorry for the confusion and thanks for helping.
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[asterisk-users] Asterisk semi-hangs
Hello, (Note: again, I'm asking for experiences/suggestions, because we have a problem in some environment that it's quite difficult to debug, test, etc.) I have seen, during last year, that some Asterisk has hangup up. I mean, not crashing, we could access to Asterisk console, but phones couldn't call. Doing dial [EMAIL PROTECTED] was not doing/showing anything, even with verbose setted up as 99. This was in Asterisk 1.2 Now, in a couple of Asterisk 1.4 boxes (I think that Asterisk 1.4.19.2?), this has happened again. Not often, every some months maybe, etc. One installation is only using VoIP, so I don't think that it's a Zaptel, Misdn, etc. problem. Even typing stop now nothing is happening, killall asterisk either, only killall -9 asterisk. We use a standard installation (no extra modules, etc.) Debian distribution, our Kernel. Has anyone some experience like this? Anything to tune? or something to see? (logs doesn't say a lot, and hard disk is not full :-) ) We would like to avoid that this is happening again, understand why is happening, etc. Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
Hi, On Apr/23/2008, Steve Totaro wrote: On Tue, Apr 22, 2008 at 7:10 AM, Carles Pina i Estany [EMAIL PROTECTED] wrote: Hello, We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1 card, 3 SPANs configured and OK and one SPAN unconfigured. In our tests it works fine, but when it has a big laod of calls (say, from 40 to 60) we have quality problems: some calls has the sound cut-off (during the call, voice was not stable) The IRQ card is alone, CPU load was not high, network was fine for sure. This server is receiving the calls from SIP channels and routing to the primaries. It's a HP server, multicore, multiCPU. I'm wondering if someone has had these kind of problems (quality problems, sound cut off) with 40 and 60 calls but not with 2 or 3, using Digium cards. Bit later I will call to Digium but I thought that here there is lot of people with lot of experience with these cards. Thank you, Just curious, are you recording these calls because that is around the I/O threshold for audio issues when recording all calls. no, we are not recording calls. Load average is very empty. We are in contact with Spanish Digium partner... Also, you say no network issues but what is the rating of your switches PPS (often overlooked for speed such as 100mb or 1000mb)? 100 Mbps, enough for 50 - 60 calls Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quality problems with ISDN PRI
Hello, We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1 card, 3 SPANs configured and OK and one SPAN unconfigured. In our tests it works fine, but when it has a big laod of calls (say, from 40 to 60) we have quality problems: some calls has the sound cut-off (during the call, voice was not stable) The IRQ card is alone, CPU load was not high, network was fine for sure. This server is receiving the calls from SIP channels and routing to the primaries. It's a HP server, multicore, multiCPU. I'm wondering if someone has had these kind of problems (quality problems, sound cut off) with 40 and 60 calls but not with 2 or 3, using Digium cards. Bit later I will call to Digium but I thought that here there is lot of people with lot of experience with these cards. Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] status line header
Hello,
I need to read the Status-Line (I need to know if it's 603, 503, 404)
after a Dial. I have tried:
exten = s,2,Dial(SIP/[EMAIL PROTECTED],,tTwW)
exten = s,3,Set(t=${SIP_HEADER(Status-Line)})
But t is empty
I have also tried:
exten = s,5,Verbose(*** STATUS: ${DIALSTATUS})
exten = s,6,Verbose(*** STATUS2: ${HANGUPCAUSE})
DIALSTATUS: I usually get Congestion (but I know from sniffer that
sometime sStatus-Line is 404, sometmies 603 and sometimes 503).
HANGUPCAUSE: I have used in ISDN but here is always 0.
How could I get the Status-Line code?
Thank you very much,
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[asterisk-users] G729 transcoding and clicking
Hello, We have an Asterisk server receiving calls using G711 (ulaw). This server has rerouters de calls to other server using G729 (we bought the codecs, installed, sip show channels shows the codec properly, etc.) Using G729, there is a click while talking. Well, more than a click it seems that voice is missing during some ms (maybe 100 ms?) Using G711 we don't have any click. Where we could watch for it? Is it possible to add some Jitter buffer? Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X-Lite Softphone keeps de-registering?
Hello, On Feb/01/2008, Doug wrote: The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? You can also use OpenVPN (using port 80, I don't know if it's possible to use TCP, I think so). So all SIP and RTP traffic would use it. -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Billing
Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Hello, On Jan/24/2008, Paul Hales wrote: http://www.transnexus.com/White% 20Papers/asterisk_V1-4-11_performance.htm It was the bottom news item on voip-info.org - I was worried I would have to really search for it! and I guess that transcoding benchmark could increase to non-transcoding calls using this card: http://digiumcards.com/digium_tc400b_transcoder_card_g729_g7231.html no? -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk scalability
Hello, I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). But because Asterisk is only one process, this process is always executed in the same CPU. So we can have a 8 Cores server, with one Core running Asterisk, another Core running operating system stuff/other small daemons and 6 idle cores. Is this correct? Why not? If this is correct, increasing CPU number of Asterisk server box would not increase the performance. I don't see any other process that could use other Cores (like transcoding processes, executing dialplan, etc.) Thank you for your information, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk scalability
Hello, On Jan/23/2008, Ryan Burke wrote: I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). Asterisk is one process, but as you mentioned multi-threaded as well. Because it is multi-threaded it can run on multiple cores/CPU's at a time. I don't know the internals of Asterisk that well so I can't site specific examples, but I know that there are some scalability bottlenecks people are looking at, specifically with the IAX protocol and how the threads send/receive packets. thanks for information. To give some more details, is we execute: ps auxwm We can see that Asterisk is using quite many threads (33 threads in a mainly new Asterisk installation) I'm sure that an Asterisk developer can chime in and give several examples of how Asterisk uses its threads to increase scalability. That said, there will be a point where the number of core/CPU's won't be the bottleneck so adding more won't help anything. Yes, I see that it uses threads. I wonder some other data like which is the limit that core/CPU's are correctly used (or usefull used). Thanks again Ryan, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Reason
Hello, I'm sniffing traffic between Asterisk and a Softswitch. I see that, in Decline SIP packages, there is a header called Reason and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1.4 and 1.2, but if it's only with Asterisk 1.4 will not be a big problem. Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Reason
Hello, On Jan/15/2008, Johansson Olle E wrote: I'm sniffing traffic between Asterisk and a Softswitch. I see that, in Decline SIP packages, there is a header called Reason and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1.4 and 1.2, but if it's only with Asterisk 1.4 will not be a big problem. There's currently no way to access that header in Asterisk. Then... next days will be time to code :-D Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Reason
Hello, On Jan/15/2008, Steve Langstaff wrote: Sent: 15 January 2008 15:23 by Johansson Olle E 15 jan 2008 kl. 15.39 skrev Steve Langstaff: Won't SIP_HEADER(reason) do that for you? No, that's only works on the INVITE that opens the dialog. The reason header comes in a reply. Thanks Olle. At least no one else saw my foolishness :) I saw :-D and I spent some minutes sniffing the SIP conversation. Yes, I cannot access to reason header but I can access to From, as Johansson said :-) Beside code, any other way how to do it? -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] channels to destroy
Hello, On Nov/19/2007, Johansson Olle E wrote: 16 nov 2007 kl. 14.06 skrev Carles Pina i Estany: In a couple of Asterisks, after type sip show channels we have a lot of these: IP_PEER dst_number something00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE [...] Is it normal? How we can remove it? Depending on the traffic on your server and whether they disappear finally after a while or hang forever, it may be a bug. Please try with the latest 1.2 version, since we spent a lot of time fixing these kind of issues earlier this year. Or even better, take time to update to version 1.4, since 1.2 is not maintained any more. ok, I will try but it will take some time. Thanks for your answer :-) Anyway, one more question: this BYE channels that doesn't disappear, can cause any problem? If the problem still persists in 1.4, please file a bug report and we'll start working on it. I will -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] not sending bye
Hello, We are using this Asterisk: 1.2.14-BRIstuffed-0.3.0-PRE-1y Everything works fine but we have an issue (not often, but one call every some hundreds) I sniffed the communication between phone, Asterisk and softswitch. I can see that Asterisk receives a Cancel from phone but Asterisk never sends a Cancel to Softswitch. This makes us some problems: billing system doesn't allow next call because there is a call limit (1 per extension), etc. Why Asterisk receives Cancel and never sends Cancel? But this happends only sometimes, not always. Yes, as soon as I get the chance I will update this Asterisk. But somebody could tell me why this is happening? I browsed in internet to find some bugreport with same behaviour without any luck. I would like to find some bug report with same problem and fine that is fixed for next Asterisk version :-) Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channels to destroy
Hello, In a couple of Asterisks, after type sip show channels we have a lot of these: IP_PEER dst_number something00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say a lot I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy, zttest
Hello, On Nov/11/2007, Tzafrir Cohen wrote: On Sun, Nov 11, 2007 at 08:51:40PM +0100, Carles Pina i Estany wrote: I also tried using bristuff 0.3y, 0.3s, etc. (is it 0.3 bristuff when Asterisk is 1.2.X?). Always without any result :-( Latest bristuff for 1.2 is y-k . See http://bristuff.org/ . However the bristuff Zaptel patch has no effect on ztdummy or Zaptel timing. I also tried y-k (last week) with no results in that machine. That computer is a Dell PowerEdge 860. I just checked now in other machine (standard Pentium 4) and just loading ztdummy and running zttest is working. In other machine, running ztdummy and zttest too. I don't know why this PowerEdge 860 is not working in that way :-( -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy and BackGround
Hello, On Nov/02/2007, Atis Lezdins wrote: On 11/2/07, Tony Plack [EMAIL PROTECTED] wrote: We are going to implement MeetMe, but this should still work right? I had similar issues with 1.4.12 just one time (also topmost zaptel at are you using Dell PowerEdge servers? -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy, zttest
Hello, On Nov/12/2007, Tony Plack wrote: Debian as well as everyone else 2.6.18-5 Zaptel is branch/1.4 latest. The issue is not with Zaptel though...IMHO. If you look at /proc/driver/rtc, I find: periodic_IRQ: no If you notice, there is no periodic_IRQ. That is the issue. To me (and If I correctly remember, I have this in yes :-| the kernels reported here) it is a Debian kernel problem. Here working with Debian, in a Dell server. I have found some references about Dell and RTC. If I turn off ACPI, Zaptel works, but the box performance is awful because Not here :-( other things, which depend on ACPI do not have interrupts. I have read some places that if you have the Hard Drive Suppend in the BIOS enabled, you will get this situation. However, I would check the /proc/driver/rtc to see that you have periodic_IRQ. I will change BIOS options and other things on Wednesday, since we don't have right now physical access to this server. In other boxes ztdummy worked like a charm... -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy, zttest
Hi, On Nov/09/2007, Tony Plack wrote: Try setting acpi=off in your boot options for the kernel. Before read your mail, I did noacpi (I guess that is the same, /proc/interrupts file changed). But without any luck. I also tried noht (no hyper threading, I think...) -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy, zttest
Hello, On Nov/10/2007, Tzafrir Cohen wrote: On Fri, Nov 09, 2007 at 04:59:37PM -0600, Tony Plack wrote: The thing is that this works, but The performance of the box becomes really bad. It seems that the problem, at least in my case is that the HPET timer from the cpu does not get an IRQ for the RTC. Does anyone else have a solution for this issue where the RTC does not get an interrupt when HPET is turned on? What kerenl version? What version of Zaptel? I know that you asked to Tony Plack, but maybe you can help me (we have same problem, it seems). I tried with last Asterisk and Zaptel, kernel version 2.6.18-4 (Debian Etch default kernel). I also tried using bristuff 0.3y, 0.3s, etc. (is it 0.3 bristuff when Asterisk is 1.2.X?). Always without any result :-( I hope that tomorrow I will have time to reproduce the same scenario with other computer. I will tell the results. Sorry to take so long to answer. Thursday and Friday I was not in office and I didn't have time... (well, I'm not in office now but I'm interested :-D) -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
Hello, On Nov/06/2007, [EMAIL PROTECTED] wrote: Just remember if you don't have any Zaptel cards you are going to have to use ztdummy to run app_meetme. Ztdummy essentially requires Linux 2.6, which you should be using anyways. yes, and this is the reason that we have setted up a new server (we didn't want to play with usual asterisk server :-) ) I found this problem :-) Thanks for advise! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy, zttest
Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than other machine that is working. CPU is a HP, Pentium 4 (same than other machine), even I loaded the same bristuff than that machine (who doesn't have any specific hardware, now). I couln't make zttest (well, ztdummy) to run. I tried different versions of bristuff+asterisk, I also tried to load and not load zaptel, qozap. Nothing. I got an rtc Warning message, something like some interruptions has been lost at 1024Hz (aprox.). Any clue where to check? USB modules are the same than other machine... We was completely confused about it (how to fix, I mean). Thanks! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe CPU resources
Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe CPU resources
Hello, First of all: also thanks to Doug Lytle and Steve Edwards. Just answering one time to all of you. I had the feeling that this computer, for 15 Meetme users, was more than enough... but we wanted to avoid any last-minute surprises! Now we are more sure that everything will work fine. Ah yes, we will use VoIP, without transcoding (I hope!), without Digium Timer card (but I will check, just in case we need it) On Nov/06/2007, Tony Mountifield wrote: In article [EMAIL PROTECTED], Carles Pina i Estany [EMAIL PROTECTED] wrote: It will depend on whether you are using VoIP or a PRI card. I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT), 1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences with up to 90 participants. I would expect them easily to handle the full Wow, 90 participants. Do you use just MeetMe in Asterisk? Just for curiosity: All of them can talk to conference? or only some of them? I thought about it, and for me, 90 open microphone participants looks like some white noise :-) Not tried here... just wondering how do you do. Thanks! -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] whisper chanspy in asterisk 1.2
Hello, I would like to have whisper channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to join two calls: one to use a private whisper and other one to use a normal chanspy. Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2)
Hello, On Mar/30/2006, Fran wrote: There are 2 ways in spain to subscribe SMS with a fixed-device (with Telefónica) -Activation Button: The device send an administrative message to subscribe SMS. There other administrative messages such as: unsubscribe, list managing, send to a list, etc... Ok, we have done it... to send SMS :-) (I think that we have problems receiving, we will call to Telefonica again...) -1rst Message sent: with your 1rst message u subscribe SMS. When a fixed-device subscribe SMS use or UBS1 or UBS2. This info is stored. Even a subscribe might have 5 fixed-devices: mixed UBS1, UBS2, one of them SMS enabled, other disabled...each subscriber device is diferent or might be different in service subscribed and protocol. ok... We have an UBS1 device for our test (its a DOMO). Most DOMOs use UBS1 but not all. I can imagine that UBS1 is Protocol 1, the same implemented in app_sms.c in Asterisk? So I shouldn't have problems sending SMS from Asterisk? Have you sent any SMS using Asterisk? I am using SMSC 900716800, without success. Thanks for all information, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS in Spain (it seems Protocol 2)
Hello,
On Mar/30/2006, Fran wrote:
I guess Protocol 1 is UBS1. I think it should be.
ok, me too...
No, i have never tested Asterisk sending messages.
We have tested some fixed devices (UBS1, UBS2 Domo type)
I have only checked Domo phone, but I don't know which protocol it is using.
Julian, from Asterisk-es (and he is here too) sent me some time ago this
link:
http://www.rtx.dk/Files/Filer/tekniske%20artikler/SMStransmissionwithinthePSTN.pdf
Maybe it is not updated, in topic about Protocol 1 and 2...
The UBS1 SMS Service is 900716800
Ok, I am using this one.
What error do u have? Timeouts? etc?
Well, I am doing this file:
Channel: Zap/1/900716800
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: smsdial
Priority: 1
Callerid: hola phone_of_FXO_card
Extension: phone_of_recipient
In extensions.conf I have this information:
[smsdial]
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup
(it is included from general section, etc.)
When I copy .call file to /var/spool/asterisk/outgoing, in Asterisk
console appears:
*CLI -- Attempting call on Zap/1/900716800 for [EMAIL PROTECTED]:1 (Retry
1)
Channel Zap/1-1 was answered.
-- Executing SMS(Zap/1-1, FXO_phone||phone_of_recipient|hola) in new
stack
-- Executing SMS(Zap/1-1, FXO_phone) in new stack
Mar 30 17:55:39 WARNING[11371]: chan_sip.c:9601 handle_response_register: Got
200 OK on REGISTER that isn't a register
== Spawn extension (smsdial, recipient_phone, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Mar 30 17:56:27 NOTICE[11380]: pbx_spool.c:280 attempt_thread: Call completed
to Zap/1/900716800
If I change 900716800 phone to France SMSC phone (0033809101000), then
it appears:
*CLI -- Attempting call on Zap/1/0033809101000 for [EMAIL PROTECTED]:1
(Retry 1)
Channel Zap/1-1 was answered.
-- Executing SMS(Zap/1-1, from_phone||to_phone|hola) in new stack
-- Executing SMS(Zap/1-1, from_phone) in new stack
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
== Spawn extension (smsdial, 600512220, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Mar 30 17:59:08 NOTICE[11403]: pbx_spool.c:280 attempt_thread: Call completed
to Zap/1/0033809101000
I rode that it should appear TX and RX lines (of course). SMS is not sent,
but maybe France SMSC is checking something (like I am not customer of there
:-) )
I don't have big knowledge about Asterisk. Maybe it is other stupid thing,
and not protocols issues...
--
Carles Pina i EstanyGPG id: 0x8CBDAE64
http://pinux.info Manresa - Barcelona
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[Asterisk-Users] SMS in Spain (it seems Protocol 2)
Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems, Spain or Italy?) If nobody is able to send, is there more people interested on it? Or any project/person/firm trying to send SMS using Protocol 2? Thank you very much, PD: some guy from Asterisk-es said to me that it seems that Telefonica wants to implement Protocol 1 too... but I don't have any information about deadlines, etc... -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calling to sip provider
Hello, I am new user of Asterisk. Yesterday I was trying to call from softphone to Asterisk, and that Asterisk routes this call to sipphone.com provider. I have found information on internet about how to register to sipphone and it seems that I have done. sip show status (or similar command) in CLI was showing me that I was registered. To call was not working, and on Asterisk's logs appeared: -- == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. -- Registered SIP '200' at 192.168.1.121 port 5060 expires 900 -- Saved useragent Linphone-1.2.0/eXosip for peer 200 -- Got SIP response 481 Subcription Does Not Exist back from 192.168.1.121 -- Executing SetCallerID(SIP/200-0e5a, Name 17476304480) in new stack -- Executing Dial(SIP/200-0e5a, SIP/[EMAIL PROTECTED]|20|r) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 500 I'm terribly sorry, server error occured (1/SL) back from 198.65.166.131 -- SIP/proxy01.sipphone.com-8a47 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/200-0e5a' status is 'CONGESTION' -- Calling using linphone or other softphones was working, so it is not circuit-busy error. I tried lot of configurations (in sip.conf and extensions.conf). Call is getting the correct route, but connection it is not working. Asterisk is behind NAT, without any redirected port. I was using externip and nat directives in configuration file. I think that I shouldn't need redirected ports because I was trying to call, not to receive calls. And NAT problem should be that I can listen but not talk (or vice-versa...) Any idea about what I can check? Any suggestion? Tahnk you very much, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
