Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
On Mon, Apr 27, 2009 at 07:37:05AM +0100, --[ UxBoD ]-- wrote: > Hi, > > Built a new server at the weekend and install Asterisk 1.6.0.9 and > IAX and SIP work great :) The one problem I am having is getting the > OpenVox (TDM400 type card) to work. It is successfully identified > using WCTDM kernel module and dahdi_scan picks it up just fine. > The issue is when I try and setup dahdi_channel.conf as it fails > everytime. What is dahdi_channel.conf ? What is the output of: lsdahdi asterisk -rx 'dahdi show channels' > When running asterisk -r I see the port pick up the ring but the > it shows a RED Alarm failure. RED alarm: nothing connected to the port (applies to FXO as well as to digital ports - E1, T1, J1, BRI). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] [UK SPECIFIC] DAHDI and a OpenVox Card
Hi, Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP work great :) The one problem I am having is getting the OpenVox (TDM400 type card) to work. It is successfully identified using WCTDM kernel module and dahdi_scan picks it up just fine. The issue is when I try and setup dahdi_channel.conf as it fails everytime. When running asterisk -r I see the port pick up the ring but the it shows a RED Alarm failure. Does anybody have a working configuration they would be happy to share with me please ? Best Regards, -- SplatNIX IT Services :: Innovation through collaboration ___ -- 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] AMD Not Working
Hi,
Thanks for your reply.
I have tried as you suggested, I does not even come upto NoOp()
It hangups after AMD.
I have decreased the silence threshold from 256 to 100 and 50.
below is the log.
-- Executing Answer("SIP/sip-38ea", "") in new stack
-- Executing AMD("SIP/sip-38ea", "") in new stack
-- AMD: SIP/sip-38ea (null) (null) (Fmt: 4)
Apr 27 00:14:25 NOTICE[20035]: app_amd.c:134 isAnsweringMachine: AMD using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [50]
-- AMD: HANGUP
vm3*CLI>
Any help is highly appreciated.
Thanks.
On Mon, Apr 27, 2009 at 3:55 AM, Matt Riddell wrote:
> On 25/04/2009 4:29 p.m., Sam Hawkin wrote:
> > Hi,
> >
> > Thanks for your reply
> >
> >
> > I have tried the HUMAN as you suggested , but still my problem does not
> > get solved.
> > I am answering the call and then running the amd.
> > Below is the log.
>
> Few things.
>
> 1. Put an answer before the AMD line.
> 2. Put a NoOp(Status: ${AMDSTATUS} Cause: ${AMDCAUSE}) after
> 3. Decrease the silence threshold
>
> --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
>
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>
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Re: [asterisk-users] function originate
As what you said, it is very difficult to control if meetme is created for each call. Playing a message after party A answers if a choice but party A will still need to hear ring after the message. She may still feel weird. Just want to know the purpose of parameter async. Can anyone tell me how to use and in which situation to use it? On Fri, Apr 24, 2009 at 5:06 PM, Geraint Lee wrote: > You could use 2 originate commands and connect both of them to a meetme > room? > > But surely what you're trying to do is going to confuse the person anyway if > they don't hear anyone when they answer? > > Wouldn't it just be better to play a message after party a answers and then > start ringing party b so that party a knows what's going on? > > 2009/4/24 Rilawich Ango >> >> Hi, >> Feature originate can be used make call thro' the web. There is a >> parameter ,Async, in it. I set it to true but there is no effect. >> Actually, I want to do the following. >> >> What I know the function originate is: >> originate call ---> party A >> party A rings >> party A answers call >> party B rings, party A still hear ring >> party B answers and A & B connected. >> party A will feel weird when she will still hear ring after answering >> a call until party B answers it. >> >> Below is what I want to do: >> originate call ---> party A >> party A rings >> party B rings >> party A answers call >> A & B connected. >> >> ___ >> -- 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 > > > ___ > -- 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 > ___ -- 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] Video Conference Software (Open Source)
I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks ___ -- 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] file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format
On Sun, Apr 26, 2009 at 1:28 PM, jonas kellens wrote:
> part of extensions.conf:
>
> *exten => 11,1,Answer()*
> *exten => 11,n,NoOp(CallerID : ${CALLERID(all)})*
> *exten => 11,n,Playback(/tmp/welkom-tcs.alaw)*
> *exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)*
> *; wordt doorgerouteerd naar context open, maar indien gesloten :*
> *exten => 11,n,NoOp(Oproep tijdens winkel gesloten)*
> *exten => 11,n,Playback(/tmp/winkel-gesloten.alaw)*
> *exten => 11,n,Hangup()*
>
> *; Record voice file to /tmp directory*
> *exten => 80,1,Wait(2) ; Call 80 to Record new Sound Files*
> *exten => 80,2,Record(/tmp/asterisk-recording:alaw) ; Press # to stop
> recording*
> *exten => 80,3,Wait(2)*
> *exten => 80,4,Playback(/tmp/asterisk-recording) ; Listen to your voice*
> *exten => 80,5,wait(2)*
> *exten => 80,6,Hangup*
>
> When I call from one of my internal SIP-clients to let Asterisk play to me
> the file "/tmp/winkel-gesloten.alaw", which I recorded in alaw-format with
> the RECORD()-application, I get the following on the Asterisk CLI :
>
> *-- Executing [...@intern:1] Answer("SIP/GXP1200-086d6b88", "") in new
> stack*
> *-- Executing [...@intern:2] NoOp("SIP/GXP1200-086d6b88", "CallerID :
> "callerid?" <51>") in new stack*
> *-- Executing [...@intern:3] Playback("SIP/GXP1200-086d6b88",
> "/tmp/welkom-tcs.alaw") in new stack*
> *[Apr 26 20:20:17] WARNING[3449]: file.c:655 ast_openstream_full: File
> /tmp/welkom-tcs.alaw does not exist in any format*
> *[Apr 26 20:20:17] WARNING[3449]: file.c:954 ast_streamfile: Unable to
> open /tmp/welkom-tcs.alaw (format 0x8 (alaw)): No such file or directory*
> *[Apr 26 20:20:17] WARNING[3449]: app_playback.c:439 playback_exec:
> ast_streamfile failed on SIP/GXP1200-086d6b88 for /tmp/welkom-tcs.alaw*
> *-- Executing [...@intern:4] GotoIfTime("SIP/GXP1200-086d6b88",
> "09:00-17:59|mon-fri|*|*?open|s|1") in new stack*
> *-- Executing [...@intern:5] NoOp("SIP/GXP1200-086d6b88", "Oproep
> tijdens winkel gesloten") in new stack*
> *-- Executing [...@intern:6] Playback("SIP/GXP1200-086d6b88",
> "/tmp/winkel-gesloten.alaw") in new stack*
> *[Apr 26 20:20:17] WARNING[3449]: file.c:655 ast_openstream_full: File
> /tmp/winkel-gesloten.alaw does not exist in any format*
> *[Apr 26 20:20:17] WARNING[3449]: file.c:954 ast_streamfile: Unable to
> open /tmp/winkel-gesloten.alaw (format 0x8 (alaw)): No such file or
> directory*
> *[Apr 26 20:20:17] WARNING[3449]: app_playback.c:439 playback_exec:
> ast_streamfile failed on SIP/GXP1200-086d6b88 for /tmp/winkel-gesloten.alaw
> *
> *-- Executing [...@intern:7] Hangup("SIP/GXP1200-086d6b88", "") in new
> stack*
> * == Spawn extension (intern, 11, 7) exited non-zero on
> 'SIP/GXP1200-086d6b88'*
>
> I've recorded the soundfile in alaw-format. The preferred format of my
> SIP-client and Asterisk is alaw. No mather where I put the sound file,
> Asterisk will not play it.
> I've put it in /var/lib/asterisk/sounds/ and in /tmp... but no succes.
> How come ?
>
>
If you check the info for the playback application, you'll find that it asks
you to not include the extension of the sound file. The extension is
appended to the filename based on the least cost of allowed codecs.
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Re: [asterisk-users] Digium fax force T38?
On Mon, 27 Apr 2009 14:37:23 David Backeberg wrote: > On Sun, Apr 26, 2009 at 8:33 AM, Michael wrote: > > Is it possible to force T38 for all invocations ReceiveFAX() ? > > If it's not T.38, it should instead be audio over G.711 or similar > codec. Are your faxes going through as audio? If not, that's a strong > indication that you have a SIP configuration issue you need to work > through before you worry about T.38. > > Are you doing something with a fancy codec like G.729? If so, stop. > You can't fax over G.729. > > You should be able to modify your dialplan to record your ReceiveFax() > session and hear exactly what is failing on your audio fax if your > audio fax is failing. Thank you for your contribution, unfortunately this problem is not going to be resolved through the normal elementary issues. I am well aware of the need to use T38, and the codec is alaw. I have already covered my bases in advance of coming on this list making my statement- 1. Hardware is fine - CW works 2. Firewall rules are fine - CW works 3. Link is fine - It's fibre 4. Provider supports T38 Further in response to Steve's comment - I have contacted my up line provider and they have advised me that they use Asterisk at their end. I am still waiting for advice as to 'what' T38 fax solution they are using, though at this point it's looking less likely that it is compatibility issues. Michael ___ -- 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] Digium fax force T38?
On Sun, Apr 26, 2009 at 8:33 AM, Michael wrote: > Is it possible to force T38 for all invocations ReceiveFAX() ? If it's not T.38, it should instead be audio over G.711 or similar codec. Are your faxes going through as audio? If not, that's a strong indication that you have a SIP configuration issue you need to work through before you worry about T.38. Are you doing something with a fancy codec like G.729? If so, stop. You can't fax over G.729. You should be able to modify your dialplan to record your ReceiveFax() session and hear exactly what is failing on your audio fax if your audio fax is failing. ___ -- 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] O/T Re: Digium fax failing
On Mon, 27 Apr 2009 04:31:14 you wrote: > Michael wrote: > > Anyway this is a great example of why MICROSOFT is worth billions, and > > Linux has to be given away. Not because Microsoft is L33T but because the > > majority of the stuff sold for Windows works out of the box. > > For what it's worth, their fax software doesn't work very well out of > the box. :-) > > Lee. True, a lot of MS applications do leave something to be desired which is why there is so many 3rd party app vendors for Windows. In terms of the usability aspect they have got this right. Internet Exploder still sucks but the Windows o/s itself is as stable as any Linux system. Free Open Source Software has been trying to gain a hold in the commercial market for some time now. Despite some small achievements, I see no evidence that it has made any major inroads. Asterisk is one example of software that has *major* potential in this area, but until this issue and many others are dealt with, it won't because it is also facing major competition from Windows equivalents from 3rd party vendors like 3CX. In response to you personally Lee, I do use the Hylafax+ software that you have been instrumental in the project, and I rate this as one of the better prospects and better supported (through the mailing list). I see this as the exception though. Michael ___ -- 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] Error, Clue to what?
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Apr 26 12:51:20] WARNING[32281]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Apr 26 12:52:56] WARNING[32284]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) We got here to 351 land [Apr 26 13:01:01] WARNING[32288]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Apr 26 14:10:01] WARNING[32294]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) We got here to VOICEPULSE land We got here to VOICEPULSE land [Apr 26 14:31:20] NOTICE[32157]: chan_iax2.c: __iax2_poke_noanswer: Peer 'brandy' is now UNREACHABLE! Time: 85 [Apr 26 14:31:30] NOTICE[32163]: chan_iax2.c:7967 socket_process: Peer 'brandy' is now REACHABLE! Time: 117 [Apr 26 15:06:07] WARNING[32300]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Apr 26 16:18:16] WARNING[32309]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) We got here to IT IS NOT NECESSARY land We got here to VOICEPULSE land We got here to VOICEPULSE land We got here to VOICEPULSE land We got here to VOICEPULSE land We got here to VOICEPULSE land We got here to VOICEPULSE land [Apr 26 17:42:41] WARNING[32324]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Apr 26 18:18:50] WARNING[32329]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) What is the (likely) cause of the above errors? It happens with little channel usage at the time. I understand that the peers were not reachable, is the dial exec full message Asterisk's message that it couldn't communicate with those peers? Thanks, Cary Fitch ___ -- 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] Digium fax failing
Michael wrote: > On Mon, 27 Apr 2009 03:40:12 you wrote: > >> Slightly off topic, but M$ is worth billions because they started in 1976 >> or so, became the de facto standard, and were pretty cutthroat in the way >> they do business. They have a profit motive and have always taken the path >> that makes them bigger with bigger profits, even to the point of fighting >> antitrust allegations. That creates billions. Linux is quite the >> opposite, and Asterisk is a friendly midpoint between profit and open >> source. YMMV. >> CF >> > > Almost every commercial customer simply want's something "that works". > > Unfortunately, with my over a decades worth of experience with Linux, if I > can't get T38 to function properly, it's a hack, not commercial grade > software. > Actually, T.38 rarely works out of the box. Its such a messed up spec. that interoperability is currently still extremely poor. Steve ___ -- 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] Hangup Detection After Originate (Asterisk Manager API)
On 24/04/2009 2:22 p.m., Saurabh Nirkhey wrote: > I have written an asterisk manager client which creates an outbound > call using Asterisk manager API's Originate action. > when the call is connected I run 3 applications on it. > 1)read a dtmf digit from user > 2)A customized application which I have written,(It plays something to user) > 3)Hangup > > If user hangs up while app 2(see above) is executing I get a 'Event Hangup' > from asterisk in my manager client . > But if app2 is over and asterisk executes Hangup (app3),It never sends > any packet to my client regarding Hangup of the call. > > I have given all permissions to manager user in manager.conf. > Can somebody help me? Maybe use the UserEvent application before calling hangup: -= Info about application 'UserEvent' =- [Synopsis] Send an arbitrary event to the manager interface [Description] UserEvent(eventname[|body]): Sends an arbitrary event to the manager interface, with an optional body representing additional arguments. The body may be specified as a | delimeted list of headers. Each additional argument will be placed on a new line in the event. The format of the event will be: Event: UserEvent UserEvent: [body] If no body is specified, only Event and UserEvent headers will be present. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] FOP and UserEvent()
On 25/04/2009 1:55 a.m., Marco Sambo wrote: > Hi all, > I try to install FOP. It's very nice. > In documentation I red that from my dial plan I can launch a popup > window with UserEvent() application. > I try to follow FOP documentation but I can't popup anything. My > structure is: > - server 1: Asterisk system > - server 2: FOP system > - client > On client I connect to FOP panel, but I don't see any popup. > > > Someone can help me to configure FOP popups and in the use of > UserEvent() application? Best to ask in the Flash Operator Panel mailing list -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] AMD Not Working
On 25/04/2009 4:29 p.m., Sam Hawkin wrote:
> Hi,
>
> Thanks for your reply
>
>
> I have tried the HUMAN as you suggested , but still my problem does not
> get solved.
> I am answering the call and then running the amd.
> Below is the log.
Few things.
1. Put an answer before the AMD line.
2. Put a NoOp(Status: ${AMDSTATUS} Cause: ${AMDCAUSE}) after
3. Decrease the silence threshold
--
Kind Regards,
Matt Riddell
Director
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Re: [asterisk-users] sipgate doesn't work with sipgate anymore
Hi, looks like I've found the solution by myself. The sipgate_out context needs the parameter insecure=invite also I missed to set the context for the dialplan. So in sip.conf using -- [sipgate_out] type=friend context=extern insecure=invite nat=yes username=1234567 fromuser=1234567 fromdomain=sipgate.de secret=secret host=sipgate.de qualify=yes -- works. Hope this information helps other people because I looked into 5 forums/links on google found 5 question on this topic but no answer ;-). Regards, Michael Michael Obster schrieb: > Hi, > > have some problem with incoming calls from sipgate. This was working in > 1.4 but in 1.6 I get a 401 Unauthorized :-(. > > Sipgate has mentioned that I have to change the type to friend, but it > is already friend, so what's wrong? > > Kind regards, > Michael > > Here is the sip.conf: > [sipgate_out] > type=friend > nat=yes > username=1234567 > fromuser=1234567 > fromdomain=sipgate.de > secret=secret > host=sipgate.de > qualify=yes > > > Here is the SIP trace: > <-> > --- (18 headers 19 lines) --- > == Using SIP RTP CoS mark 5 > Sending to 217.10.79.9 : 5060 (no NAT) > Using INVITE request as basis request - > 40fa80331421fa800e4633bd497e1...@sipgate.de > No user '015122633153' in SIP users list > Found peer 'sipgate_out' for '015122633153' from 217.10.79.9:5060 > > <--- Reliably Transmitting (NAT) to 217.10.79.9:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 217.10.79.9:5060;branch=z9hG4bK14af.633376b5.0;received=217.10.79.9 > Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK14af.633376b5.0 > Via: SIP/2.0/UDP > 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK6263330b > Via: SIP/2.0/UDP 217.10.67.8:5060;branch=z9hG4bK6263330b;rport=5060 > From: "015122633153" ;tag=as2931c3cc > To: ;tag=as795f5a0d > Call-ID: 40fa80331421fa800e4633bd497e1...@sipgate.de > CSeq: 102 INVITE > User-Agent: Asterisk PBX 1.6.0.9 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="041f4025" > Content-Length: 0 > > > <> > Scheduling destruction of SIP dialog > '40fa80331421fa800e4633bd497e1...@sipgate.de' in 6400 ms (Method: INVITE) > netmaster*CLI> > <--- SIP read from UDP://217.10.79.9:5060 ---> > ACK sip:1234...@192.168.173.2:5060 SIP/2.0 > Max-Forwards: 10 > Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK14af.633376b5.0 > Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK14af.633376b5.0 > From: "015122633153" ;tag=as2931c3cc > Call-ID: 40fa80331421fa800e4633bd497e1...@sipgate.de > To: ;tag=as795f5a0d > CSeq: 102 ACK > Content-Length: 0 > X-hint: rr-enforced > > > > ___ > -- 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 ___ -- 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] 1.6.1: "DNS error" but ping works
Probably the phone is using a "wrong" DNS entry. Try changing the Proxy address to the IP address and see if the phone registers. If so, work backwards from there. Change DNS setting to some other DNS. And, remember you have to set an IP DNS, you can't use a URL to look up a DNS. (Assuming you don't have a DNS available to start with.) Also some low end routers are a little flakey about getting a real DNS and serving as a DNS for downstream devices. It is more sure to specify YOUR DNS addresses on the phone. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Sunday, April 26, 2009 2:03 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1.6.1: "DNS error" but ping works With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121...@proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS error for registration to 1747yyyx...@proxy01.sipphone.com, trying REGISTER again (after 20 seconds) but ping works: ping proxy01.sipphone.com PING northamerica.sipphone.com (198.65.166.131) 56(84) bytes of data. 64 bytes from northamerica.sipphone.com (198.65.166.131): icmp_seq=1 ttl=52 time=96.5 ms 64 bytes from northamerica.sipphone.com (198.65.166.131): icmp_seq=2 ttl=52 time=94.4 ms Is this a bug, or could it be caused by a faulty configuration? sean ___ -- 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 ___ -- 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] 1.6.1: "DNS error" but ping works
With 1.6.1 svn: [2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout: -- Registration for '17470121...@proxy01.sipphone.com' timed out, trying again (Attempt #30) [2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'proxy01.sipphone.com' [2009-04-26 15:01:00] WARNING[1844]: chan_sip.c:10037 transmit_register: Probably a DNS error for registration to 1747yyyx...@proxy01.sipphone.com, trying REGISTER again (after 20 seconds) but ping works: ping proxy01.sipphone.com PING northamerica.sipphone.com (198.65.166.131) 56(84) bytes of data. 64 bytes from northamerica.sipphone.com (198.65.166.131): icmp_seq=1 ttl=52 time=96.5 ms 64 bytes from northamerica.sipphone.com (198.65.166.131): icmp_seq=2 ttl=52 time=94.4 ms Is this a bug, or could it be caused by a faulty configuration? sean ___ -- 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] sipgate doesn't work with sipgate anymore
Hi, have some problem with incoming calls from sipgate. This was working in 1.4 but in 1.6 I get a 401 Unauthorized :-(. Sipgate has mentioned that I have to change the type to friend, but it is already friend, so what's wrong? Kind regards, Michael Here is the sip.conf: [sipgate_out] type=friend nat=yes username=1234567 fromuser=1234567 fromdomain=sipgate.de secret=secret host=sipgate.de qualify=yes Here is the SIP trace: <-> --- (18 headers 19 lines) --- == Using SIP RTP CoS mark 5 Sending to 217.10.79.9 : 5060 (no NAT) Using INVITE request as basis request - 40fa80331421fa800e4633bd497e1...@sipgate.de No user '015122633153' in SIP users list Found peer 'sipgate_out' for '015122633153' from 217.10.79.9:5060 <--- Reliably Transmitting (NAT) to 217.10.79.9:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK14af.633376b5.0;received=217.10.79.9 Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK14af.633376b5.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK6263330b Via: SIP/2.0/UDP 217.10.67.8:5060;branch=z9hG4bK6263330b;rport=5060 From: "015122633153" ;tag=as2931c3cc To: ;tag=as795f5a0d Call-ID: 40fa80331421fa800e4633bd497e1...@sipgate.de CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="041f4025" Content-Length: 0 <> Scheduling destruction of SIP dialog '40fa80331421fa800e4633bd497e1...@sipgate.de' in 6400 ms (Method: INVITE) netmaster*CLI> <--- SIP read from UDP://217.10.79.9:5060 ---> ACK sip:1234...@192.168.173.2:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK14af.633376b5.0 Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK14af.633376b5.0 From: "015122633153" ;tag=as2931c3cc Call-ID: 40fa80331421fa800e4633bd497e1...@sipgate.de To: ;tag=as795f5a0d CSeq: 102 ACK Content-Length: 0 X-hint: rr-enforced ___ -- 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] file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format
part of extensions.conf:
exten => 11,1,Answer()
exten => 11,n,NoOp(CallerID : ${CALLERID(all)})
exten => 11,n,Playback(/tmp/welkom-tcs.alaw)
exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten => 11,n,NoOp(Oproep tijdens winkel gesloten)
exten => 11,n,Playback(/tmp/winkel-gesloten.alaw)
exten => 11,n,Hangup()
; Record voice file to /tmp directory
exten => 80,1,Wait(2) ; Call 80 to Record new Sound Files
exten => 80,2,Record(/tmp/asterisk-recording:alaw) ; Press # to stop
recording
exten => 80,3,Wait(2)
exten => 80,4,Playback(/tmp/asterisk-recording) ; Listen to your voice
exten => 80,5,wait(2)
exten => 80,6,Hangup
When I call from one of my internal SIP-clients to let Asterisk play to
me the file "/tmp/winkel-gesloten.alaw", which I recorded in alaw-format
with the RECORD()-application, I get the following on the Asterisk CLI :
-- Executing [...@intern:1] Answer("SIP/GXP1200-086d6b88", "") in new
stack
-- Executing [...@intern:2] NoOp("SIP/GXP1200-086d6b88", "CallerID :
"callerid?" <51>") in new stack
-- Executing [...@intern:3] Playback("SIP/GXP1200-086d6b88",
"/tmp/welkom-tcs.alaw") in new stack
[Apr 26 20:20:17] WARNING[3449]: file.c:655 ast_openstream_full:
File /tmp/welkom-tcs.alaw does not exist in any format
[Apr 26 20:20:17] WARNING[3449]: file.c:954 ast_streamfile: Unable to
open /tmp/welkom-tcs.alaw (format 0x8 (alaw)): No such file or directory
[Apr 26 20:20:17] WARNING[3449]: app_playback.c:439 playback_exec:
ast_streamfile failed on SIP/GXP1200-086d6b88 for /tmp/welkom-tcs.alaw
-- Executing [...@intern:4] GotoIfTime("SIP/GXP1200-086d6b88",
"09:00-17:59|mon-fri|*|*?open|s|1") in new stack
-- Executing [...@intern:5] NoOp("SIP/GXP1200-086d6b88", "Oproep
tijdens winkel gesloten") in new stack
-- Executing [...@intern:6] Playback("SIP/GXP1200-086d6b88",
"/tmp/winkel-gesloten.alaw") in new stack
[Apr 26 20:20:17] WARNING[3449]: file.c:655 ast_openstream_full:
File /tmp/winkel-gesloten.alaw does not exist in any format
[Apr 26 20:20:17] WARNING[3449]: file.c:954 ast_streamfile: Unable to
open /tmp/winkel-gesloten.alaw (format 0x8 (alaw)): No such file or
directory
[Apr 26 20:20:17] WARNING[3449]: app_playback.c:439 playback_exec:
ast_streamfile failed on SIP/GXP1200-086d6b88
for /tmp/winkel-gesloten.alaw
-- Executing [...@intern:7] Hangup("SIP/GXP1200-086d6b88", "") in new
stack
== Spawn extension (intern, 11, 7) exited non-zero on
'SIP/GXP1200-086d6b88'
I've recorded the soundfile in alaw-format. The preferred format of my
SIP-client and Asterisk is alaw. No mather where I put the sound file,
Asterisk will not play it.
I've put it in /var/lib/asterisk/sounds/ and in /tmp... but no succes.
How come ?
Thanks for the help !
Greetingz,
Jonas.
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Re: [asterisk-users] Digium fax failing
Michael wrote: > On Mon, 27 Apr 2009 03:40:12 you wrote: > >> Slightly off topic, but M$ is worth billions because they started in 1976 >> or so, became the de facto standard, and were pretty cutthroat in the way >> they do business. They have a profit motive and have always taken the path >> that makes them bigger with bigger profits, even to the point of fighting >> antitrust allegations. That creates billions. Linux is quite the >> opposite, and Asterisk is a friendly midpoint between profit and open >> source. YMMV. >> CF >> > > Almost every commercial customer simply want's something "that works". > > Unfortunately, with my over a decades worth of experience with Linux, if I > can't get T38 to function properly, it's a hack, not commercial grade > software. It sounded to me like the T.38 functionality in Callweaver was working for you. Why did you abandon it? Lee. ___ -- 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] Digium fax failing
Michael wrote: > Anyway this is a great example of why MICROSOFT is worth billions, and Linux > has to be given away. Not because Microsoft is L33T but because the majority > of the stuff sold for Windows works out of the box. For what it's worth, their fax software doesn't work very well out of the box. :-) Lee. ___ -- 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] Digium fax failing
On Mon, 27 Apr 2009 03:40:12 you wrote: > Slightly off topic, but M$ is worth billions because they started in 1976 > or so, became the de facto standard, and were pretty cutthroat in the way > they do business. They have a profit motive and have always taken the path > that makes them bigger with bigger profits, even to the point of fighting > antitrust allegations. That creates billions. Linux is quite the > opposite, and Asterisk is a friendly midpoint between profit and open > source. YMMV. > CF Almost every commercial customer simply want's something "that works". Unfortunately, with my over a decades worth of experience with Linux, if I can't get T38 to function properly, it's a hack, not commercial grade software. Michael ___ -- 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] Digium fax failing
Slightly off topic, but M$ is worth billions because they started in 1976 or so, became the de facto standard, and were pretty cutthroat in the way they do business. They have a profit motive and have always taken the path that makes them bigger with bigger profits, even to the point of fighting antitrust allegations. That creates billions. Linux is quite the opposite, and Asterisk is a friendly midpoint between profit and open source. YMMV. CF === "Anyway this is a great example of why MICROSOFT is worth billions, and Linux has to be given away. Not because Microsoft is L33T but because the majority of the stuff sold for Windows works out of the box. Michael" ___ -- 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 ___ -- 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] 1.6.1: menuselect has problems with x86_64 ??
sean darcy schrieb: > 1.6.1 svn 190575: > > CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect > CONFIGURE_SILENT="--silent" menuselect > make[1]: Entering directory > `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' > gcc -m64 -march=native -mtune=native -floop-interchange > -floop-strip-mine -floop-block -c -o menuselect_stub.o menuselect_stub.c > gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a > /usr/bin/ld: i386 architecture of input file `menuselect.o' is > incompatible with i386:x86-64 output > /usr/bin/ld: i386 architecture of input file `strcompat.o' is > incompatible with i386:x86-64 output Did you run `make distclean`? Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Digium fax failing
On Mon, 27 Apr 2009 02:33:31 you wrote: > have a look at the dahdi or zaptel configs and search for echo In > the last 9+ years of working with * I have never had a manual, whats > it look like..? > > Andrew "lathama" Latham > > TuxTone Inc. > http://TuxTone.com > andrew.lat...@tuxtone.com And I also very clearly stated the up link was T38. Anyway this is a great example of why MICROSOFT is worth billions, and Linux has to be given away. Not because Microsoft is L33T but because the majority of the stuff sold for Windows works out of the box. Yes, I know RTFM. Which I have done. End to end a few times. It isn't the system - Callweaver can receive faxes on the same hardware + link + upline provider, however in true Linux fashion it can't send faxes Michael ___ -- 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] 1.6.1: menuselect has problems with x86_64 ??
On Sun, Apr 26, 2009 at 10:26:12AM -0400, sean darcy wrote: > 1.6.1 svn 190575: > > CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect > CONFIGURE_SILENT="--silent" menuselect > make[1]: Entering directory > `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' > gcc -m64 -march=native -mtune=native -floop-interchange > -floop-strip-mine -floop-block -c -o menuselect_stub.o menuselect_stub.c > gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a > /usr/bin/ld: i386 architecture of input file `menuselect.o' is > incompatible with i386:x86-64 output > /usr/bin/ld: i386 architecture of input file `strcompat.o' is > incompatible with i386:x86-64 output Any chance you built the same directory earlier on i386? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] 1.6.1: menuselect has problems with x86_64 ??
1.6.1 svn 190575: CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" menuselect make[1]: Entering directory `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' gcc -m64 -march=native -mtune=native -floop-interchange -floop-strip-mine -floop-block -c -o menuselect_stub.o menuselect_stub.c gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o mxml/libmxml.a /usr/bin/ld: i386 architecture of input file `menuselect.o' is incompatible with i386:x86-64 output /usr/bin/ld: i386 architecture of input file `strcompat.o' is incompatible with i386:x86-64 output sean ___ -- 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] Digium fax failing
On Mon, 27 Apr 2009 02:02:12 you wrote: > When configured with echo training (default) Asterisk attempts to > train its self on all calls. You have to tell it to either not train > on all calls or turn on fax detection. > > > Andrew "lathama" Latham How do I turn this on please? I can't find any mention of it in the Digium manual. Michael ___ -- 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] Digium fax failing
On Mon, 27 Apr 2009 01:36:25 you wrote: > Turn on fax detection so that echo training will not attempt to run on > the fax handshake... > > > Andrew "lathama" Latham > > TuxTone Inc. > http://TuxTone.com > andrew.lat...@tuxtone.com Thanks. How is this done on a T38 fax channel please? PS: I see no mention of this in the Digium guide - have they forgotten something? Michael ___ -- 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] Digium fax failing
Turn on fax detection so that echo training will not attempt to run on the fax handshake... Andrew "lathama" Latham TuxTone Inc. http://TuxTone.com andrew.lat...@tuxtone.com On Sat, Apr 25, 2009 at 10:21 PM, Michael wrote: > Sending works but on receive it keeps failing - reporting back 'training' > failure. > > I am using Asterisk 1.6 with T38. > > What should I post to the list to assist diagnoses? > > Michael > > ___ > -- 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 > ___ -- 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] Digium fax force T38?
Is it possible to force T38 for all invocations ReceiveFAX() ? Receiving fax always worked OK on Callweaver though I could put SipT38Switchover() into the dial plan. I can't with Digium fax, and it always fails at the point it decides to switch to T38. ___ -- 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] Compact, fanless appliance?
On 26 Apr 2009, at 09:17, Paul Chambers wrote: Vincent wrote: www.voip-info.org/wiki/view/Asterisk+embedded+systems Thanks Steve. I knew about this list, but I wanted to make sure there weren't other, more complete sources about the subject. So at this point, it seems like it boils down to this: Soekris PCEngines Atcom (IP01: £160+VAT) Herologic (HL-463: $259) uCpbx (235.00 EUR) AstBoxes (168.00 EUR) Gumstix HP Thinclient t5720 Thank you. EdgePBX is another option - they offer two, eight and twelve port Astfin (Asterisk on Blackfin) boxes, similar to those from uCpbx & Atcom. The FX02 and FX08 have SD slots, the FX12 has a CF slot. None have PCI slots or USB, just Ethernet. The FX02 (two port) is $150, plus $25 per module, though any compatible TDM400-style module will work. I have an FX08 with Digium and NetX86 modules in it. It's been solid for me (just a customer, no connection to the company). I'm running asterisk 1.4 on an NSLU2 , only a couple of channels and minimal transcoding, but it seems fine and stable. £80 + usb storage I built 1.4 from sources on the NSLU2 which took a while :-) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] FW: issue with sip 180 responses
Hello, It's happens around 40 calls and above … The **machine** accepts number of invites(we can see by tcpdump ) , but asterisk sees part of them (we can see by CLI log) , and when it does , asterisk accepting an invite it reply the initator. (as it should ) – but the rest of invites are just ignored. it's seems like an O/S issue, because on asterisk level I can all going correctly via logs (invite is accepted=> packet is generated=> and 180 is sent immediately to the initiator …) which tools can help me check kernel issues ? Also, tried to increase udp buffer (sysctl -w net.core.rmem_max=8388608) , but seems the problem still persists. Also , here a screenshot of a typical dump from network interface, you can clearly see what's going on. http://img7.imageshack.us/img7/6578/sip.png Thank in advanced , Nir. *C. Savinovich*** did you isolated the issue? , checked firewall , interface errors , routing , sniffed the interface…. also , why using h323 and not IAX2 ? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *C. Savinovich *Sent:* Sunday, April 19, 2009 5:02 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] issue with sip 180 responses I am having a similar issue. Asterisk does not show ringback tone and I investigated this due to it not reading sip invite 180. (or supposedly not receiving it).. My solution is that now I am using h323 (ver 1.4.19) CS ___ -- 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] Parked calls for multiple customers
carl Lougher wrote: > Ok cheers. > > Any idea when 1.6 goes stable for prod? Theoretically it already has, however as was the case with 1.4, I suggest you tread very carefully when it comes to migrating to 1.6. ___ -- 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] Compact, fanless appliance?
Vincent wrote: > www.voip-info.org/wiki/view/Asterisk+embedded+systems > > Thanks Steve. I knew about this list, but I wanted to make sure there > weren't other, more complete sources about the subject. > > So at this point, it seems like it boils down to this: > Soekris > PCEngines > Atcom (IP01: £160+VAT) > Herologic (HL-463: $259) > uCpbx (235.00 EUR) > AstBoxes (168.00 EUR) > Gumstix > HP Thinclient t5720 > > Thank you. > EdgePBX is another option - they offer two, eight and twelve port Astfin (Asterisk on Blackfin) boxes, similar to those from uCpbx & Atcom. The FX02 and FX08 have SD slots, the FX12 has a CF slot. None have PCI slots or USB, just Ethernet. The FX02 (two port) is $150, plus $25 per module, though any compatible TDM400-style module will work. I have an FX08 with Digium and NetX86 modules in it. It's been solid for me (just a customer, no connection to the company). -- Paul ___ -- 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] timing source problem
On Fri, Apr 24, 2009 at 12:09:50PM +0200, Wolfgang Pichler wrote: > hi all, > > we do have some troubles with zaptel timing source - we have a setup > with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk > does some handling - calls are leaving on digium card 1 - going to a > siemens hipath - there is some call handling - some of the calls then > are going from the hipath over a qsig line to a bosch integral PBX - > handling the rest of the calls. > > To be able to get away from the bosch system - we like to put asterisk > (1 port free on each card) in the middle of the path siemens -> bosch - > so that it will be siemens -> card 0 asterisk card 1 -> bosch. > > Currently the Siemens hipath is playing the network side - the bosch is > cpe. So the siemens hipath does provide the timing source. > > With asterisk in the middle i can not take the timing source from the > siemens link - because i have already the telco line as timing source. > But when starting it in this setup - i will get lots of "timing source > auto card 0!" messages. So i think the siemens timing is not in sync. > with the telco timing - so mixed up on asterisk with telco line as > primary timing will not work when the siemens does try to deliver > timing. > > I have not tried as /etc/zaptel.conf parameter 0 als timing parameter (0 > = be master) - but i think it wont work because the siemens wont accept > the timing from the asterisk box. > > Changing configuration of the siemens is not possible. > > So - here the questions... > - is it possible to do what i want to do ? > - do you think timing=0 in zaptel.conf will work ? > - would it be possible to connect a xorcom 2 PRI channel bank to > asterisk to handle the qsig line between the two ? Or will the xorcom > then also take the timing from the digum cards - telco lines ? I missed the part about Xorcom initially and I see nobody answered it yet. Yes, a Xorcom PRI module should be able to take fit here and get timing from the main Zaptel / DAHDI timing source on the system. IIRC, though (some? all?) Digium digital cards have a sync cable option which may be what you need here. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Parked calls for multiple customers
Ok cheers. Any idea when 1.6 goes stable for prod? - Original Message From: Mike To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, 24 April, 2009 0:54:59 Subject: Re: [asterisk-users] Parked calls for multiple customers No, but as I understand it 1.6 would have that possibility. Mike > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of carl Lougher > Sent: Thursday, April 23, 2009 4:54 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Parked calls for multiple customers > > > Hi, > > Is there any method of getting call park working on different numbers for > different customers on the same asterisk server? > Currently running asterisk 1.4.23.1 > > Cheers!! > > > > > ___ > -- 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 ___ -- 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 ___ -- 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
