[asterisk-users] unnumbered priorities
Hi: When should we use unnumbered priorities(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten = 333,1,Answer exten = 333,n,meetme(8000|cim) exten = 333,n,playback(vm-goodbye) exten = 333,n,hangup form2: [Conferencerooms] exten = 333,1,Answer exten = 333,2,meetme(8000|cim) exten = 333,3,playback(vm-goodbye) exten = 333,4,hangup I'd appreciate any help. Regards. - Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search.___ --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] enter menu
For user and administrator enter menu when *-key is pressed we should use 's' option or nothing(asterisk does it automatic). I'd appreciate any idea. Regards. - Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. ___ --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] unnumbered priorities
Am Sonntag, den 02.09.2007, 23:25 -0700 schrieb fateme fatah: Hi: When should we use unnumbered priorities(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten = 333,1,Answer exten = 333,n,meetme(8000|cim) exten = 333,n,playback(vm-goodbye) exten = 333,n,hangup form2: [Conferencerooms] exten = 333,1,Answer exten = 333,2,meetme(8000|cim) exten = 333,3,playback(vm-goodbye) exten = 333,4,hangup On one of my Asterisk 1.2 setups, I have a dialplan that mixes both - so they can coexist in the same extensions.conf. The difference is that with the n type extensions, you can easily insert a line or three without renumbering lots of lines - and searching for all those GOTOs that also need a new line number. Renumbering error-prone. An advantage of numbering is that the line order is not important, because of course Asterisk would select by number, not order - and possibly (although I did not investigate this) including _parts_ of an extension from another context might work better. All my new extensions use the n style, but I am not going to rewrite the older parts of the dialplan soon. BR Anselm ___ --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] Rechazo de llamada en triangulacion de asterisk.
Gracias Alex. Lo pondré el Lunes y te tire como van las cosas. Guillermo El Jueves, 30 de Agosto de 2007 17:52, Alex Balashov escribió: Guillermo, Me parece que la cosa aqui es que el nombre del usuario debe ser el mismo en el URI del fuente que en el el proceso de autentificacion. Traiga poner username= en la configuracion asi: On Thu, 30 Aug 2007, Guillermo Rodriguez wrote: [pbx1] name=test1 callerid=200 host=dynamic nat = yes type friend secret= test1 username=... Y diganos lo que pasa. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Account Registration Failed
Hi, I am trying to run an Asterisk (1.4.11) server on Linux Suse. The server is behind NAT. I am testing with SIP client that developed from PJSIP running on Pocket PC Windows Mobile 5.0 . The client is also behind another NAT. STUN server is implement in SIP client. When a register Message send from client to server, Asterisk receive it and reply with 100 trying msg. However, there is no reply on 200 OK from server, as it course my SIP client registration failure. 1. On the other hand, if i tested with OPENSER SIP server, registration is fine. Important details are below: sip.conf [global] nat=yes canreinvite=no localnet=192.168.1.46 externip=60.xx.xx.xx.xx [8000] type=friend secret=8000 nat=yes host=dynamic canreinvite=no How can i solve it? p/s : Network traffic capture in Ethereal are attached. ~ cobra client.cap - capture at client side ~ cobra server - capture at Asterisk server Thanks Regards kum _ Stay connected with your friends and discover new ones on Windows Live Spaces! http://spaces.live.com?mkt=en-my cobra client.cap Description: Binary data cobra server.cap Description: Binary data ___ --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] Wanted: VoIP Engineer for Warsaw!
Peoplefone AG jest firmą oferującą usługi telekomunikacyjne oparte na technologii Voice over IP (VoIP) w wyjątkowo niskich stawkach. Peoplefone jest certyfikowanym partnerem firm Siemenshttp://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1sdc_sid=1113876080;i AVM/FRITZ!Box http://www.fritz-shop.ch/. W związku z naszym dynamicznym rozwojem, do nowo tworzonej spółki w Polsce poszukujemy osoby na stanowisko: SPECJALISTA D/S VOIP Miejsce pracy: Warszawa *Wymagania:* - wykształcenie wyższe techniczne (lub w trakcie studiów), najlepiej informatyczne - praktyczna umiejętność programowania w środowisku w PHP i językach C i C++ - znajomość Linux - znajomość mySQL i PostgreSQL - znajomość zagadnień związanych z sieciami IP, UDP, TCP - znajomość narzędzi do analizy sieci np.. Ethereal - wiedza z zakresu VoIP, serwerów i konfiguracji oprzyrządowania - umiejętność rozwiązywania problemów informatycznych - znajomość języka angielskiego w mowie i piśmie w stopniu zaawansowanym - umiejętność szybkiego uczenia się - chęć podnoszenia swoich kwalifikacji zawodowych - profesjonalne i rzetelne podejście do obowiązków służbowych, umiejętność pracy w zespole, samodzielność, komunikatywność, systematyczność, terminowość - doświadczenie zawodowe mile widziane *Dodatkowe wymagania (mile widziane):* - umiejętność programowania w PERL i JAVA - znajomość Asterisk / SER / OPENSER - umiejętność konfigurowania bramek i ruterów Cisco, Patton, lub innych - znajomość protokołów SIP lub STUN - znajomość problematyki NAT I funkcjonowania outband Proxy - znajomość narzędzi monitorujących jak Cactus, Nagios, MRTG - znajomość narzędzi jak DRBD, Hearthbeat - dyspozycyjność Wszystkich zainteresowanych prosimy o przesłanie życiorysu oraz listu motywacyjnego (zawierającego zgodę na przetwarzanie danych osobowych) *do 10 września br. ** *na adres: [EMAIL PROTECTED] W temacie wiadomości wysyłanej pocztą elektroniczną prosimy umieścić nazwę stanowiska. Skontaktujemy się tylko z wybranymi kandydatami. Prosimy o dopisanie następującej klauzuli Wyrażam zgodę na przetwarzanie moich danych osobowych zawartych w mojej ofercie pracy dla potrzeb niezbędnych do realizacji procesu rekrutacji (zgodnie z ustawą z dn. 29.08.97 roku o Ochronie Danych Osobowych Dz. Ust Nr 133 poz. 883). ___ --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] unnumbered priorities
In article [EMAIL PROTECTED], Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Sonntag, den 02.09.2007, 23:25 -0700 schrieb fateme fatah: Hi: When should we use unnumbered priorities(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten = 333,1,Answer exten = 333,n,meetme(8000|cim) exten = 333,n,playback(vm-goodbye) exten = 333,n,hangup form2: [Conferencerooms] exten = 333,1,Answer exten = 333,2,meetme(8000|cim) exten = 333,3,playback(vm-goodbye) exten = 333,4,hangup On one of my Asterisk 1.2 setups, I have a dialplan that mixes both - so they can coexist in the same extensions.conf. The difference is that with the n type extensions, you can easily insert a line or three without renumbering lots of lines - and searching for all those GOTOs that also need a new line number. Renumbering error-prone. In fact the 'n' is just a shortcut for next number and is translated into an actual number when the extensions.conf file is read. If you do show dialplan, you will see all the extensions have numbers, even if you had used 'n'. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Rechazo de llamada en triangulacion de asterisk.
Hola Alex, He puesto el username y aun asi me da el fallo de autentificacion :-( Creo que lo mas facil sería cambiar el name de los telefonos.. aunque seria lo mas complicado ya que tendria que cambiar el name de todos mis clientes... y realmente no sera precisamente una solucion viable. El Jueves, 30 de Agosto de 2007 17:52, Alex Balashov escribió: Guillermo, Me parece que la cosa aqui es que el nombre del usuario debe ser el mismo en el URI del fuente que en el el proceso de autentificacion. Traiga poner username= en la configuracion asi: On Thu, 30 Aug 2007, Guillermo Rodriguez wrote: [pbx1] name=test1 callerid=200 host=dynamic nat = yes type friend secret= test1 username=... Y diganos lo que pasa. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Rechazo de llamada en triangulacion de asterisk.
Guillermo, el username deberia de ser igual al nombre del canal, es decir, si [pbx1] entonces username = pbx1 Saludos, Jonathan GF On 9/3/07, Guillermo Rodriguez [EMAIL PROTECTED] wrote: Hola Alex, He puesto el username y aun asi me da el fallo de autentificacion :-( Creo que lo mas facil sería cambiar el name de los telefonos.. aunque seria lo mas complicado ya que tendria que cambiar el name de todos mis clientes... y realmente no sera precisamente una solucion viable. El Jueves, 30 de Agosto de 2007 17:52, Alex Balashov escribió: Guillermo, Me parece que la cosa aqui es que el nombre del usuario debe ser el mismo en el URI del fuente que en el el proceso de autentificacion. Traiga poner username= en la configuracion asi: On Thu, 30 Aug 2007, Guillermo Rodriguez wrote: [pbx1] name=test1 callerid=200 host=dynamic nat = yes type friend secret= test1 username=... Y diganos lo que pasa. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Setting Callerid with chan_misdn
Hello, I am using asterisk-1.2 with chan_misdn and hava a usb-isdn adapter. Calling in and calling out works quite well, except that it is not possible to set the callerid. I have a connection with 4 telephone numbers (german ISDN), the problem is if I make a call to my mobile the basenumber is shown on my mobile as the source of the call. How can I set the phone number which is shown as the source of the call? In my dialplan I have the following line: Set(CALLERID=..) which does not work, only the basenumber is shown.. Any ideas? Regards Jan ___ --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] Dificult macro, please advise
Hi,
BRIEF RESUME:
Is there any other way to obtain the same result but being easier to
configure?? Thanks!
EXTENDED RESUME:
i've configured a, rather difficult, macro that even for me without
being documented is difficult. I ask for the help of the experts to
know if the functionality it apports can be achieved better in another
way.
What i'm trying is to enable call a channel (e.g. SIP/3) and being
able to leave a message only on box 2 while any other call to any
other user (e.g.SIP/1 or SIP/2) and leave a message only on mailbox 1.
This is inteded for a SOHO environment.
I have defined only two mailboxes, the 1'st for personal ussage and
the 2'nd for profesional usage.
The macro that now allows me to do that is the following:
EXT_CALLER=
EXT_STUDIO=SIP/3
exten = _X,1,Macro(diallocal|SIP/${EXTEN}|20,Tr)
[macro-diallocal]
exten = s,1,Dial(${ARG1}|${ARG2}|${ARG3})
exten = s,2,Set(EXT_CALLED=${ARG1})
exten =
s,3,GotoIf($[${EXT_CALLED}=${EXT_STUDIO}]?s-STUDIO-${DIALSTATUS},1:s-REST-${DIALSTATUS},1)
exten = s-STUDIO-NOANSWER,1,Voicemail(u2)
exten = s-STUDIO-NOANSWER,2,Hangup()
exten = s-STUDIO-BUSY,1,Voicemail(b2)
exten = s-STUDIO-BUSY,2,Hangup()
exten = _s-STUDIO-.,1,Hangup()
exten = s-REST-NOANSWER,1,Voicemail(u1)
exten = s-REST-NOANSWER,2,Hangup()
exten = s-REST-BUSY,1,Voicemail(b1)
exten = s-REST-BUSY,2,Hangup()
exten = _s-REST-.,1,Hangup()
Is there any other way to obtain the same result but being easier to configure??
Thanks in advance.
Best regards,
Jonathan GF
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Re: [asterisk-users] Rechazo de llamada en triangulacion de asterisk.
On Mon, 3 Sep 2007, Jonathan GF wrote: Guillermo, el username deberia de ser igual al nombre del canal, es decir, si [pbx1] entonces username = pbx1 Ay, si. Eso se me olvido, aun Jonathan tiene razon. Ya recurdo que tambien he tenido ese problema. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Rechazo de llamada en triangulacion de asterisk.
Saludos a todos. El caballo que me vendiste ya no sirve. Mi chevy es en el shop. Let's make this Spanish Day for the list. Tomorrow will be Latin Day and the day after will be Ancient Egyptian Day. Just kidding. For those in countries where today is a holiday - enjoy it. For the others - take an extra hour for lunch and a long siesta afterwards. Alex Balashov wrote: On Mon, 3 Sep 2007, Jonathan GF wrote: Guillermo, el username deberia de ser igual al nombre del canal, es decir, si [pbx1] entonces username = pbx1 Ay, si. Eso se me olvido, aun Jonathan tiene razon. Ya recurdo que tambien he tenido ese problema. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Rechazo de llamada en triangulacion de asterisk.
On Mon, 3 Sep 2007, Paul wrote: Just kidding. For those in countries where today is a holiday - enjoy it. For the others - take an extra hour for lunch and a long siesta afterwards. In the fine tradition of open-source development in today's fast-paced, ever-changing global economy of mission-critical, high-ROI 24/7 B2B deliverables, I call on folks to do that every day. :-) Results over presence -- teach your employer to value it! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Dificult macro, please advise
On 9/3/07, Jonathan GF [EMAIL PROTECTED] wrote:
Hi,
BRIEF RESUME:
Is there any other way to obtain the same result but being easier to
configure?? Thanks!
Why don't you use asterisk extension masks?
exten = _3,1,Macro(dial-studio|SIP/${EXTEN}|20|Tr)
exten = _X,1,Macro(dial-rest|SIP/${EXTEN}|20|Tr)
Regards,
Atis
EXTENDED RESUME:
i've configured a, rather difficult, macro that even for me without
being documented is difficult. I ask for the help of the experts to
know if the functionality it apports can be achieved better in another
way.
What i'm trying is to enable call a channel (e.g. SIP/3) and being
able to leave a message only on box 2 while any other call to any
other user (e.g.SIP/1 or SIP/2) and leave a message only on mailbox 1.
This is inteded for a SOHO environment.
I have defined only two mailboxes, the 1'st for personal ussage and
the 2'nd for profesional usage.
The macro that now allows me to do that is the following:
EXT_CALLER=
EXT_STUDIO=SIP/3
exten = _X,1,Macro(diallocal|SIP/${EXTEN}|20,Tr)
[macro-diallocal]
exten = s,1,Dial(${ARG1}|${ARG2}|${ARG3})
exten = s,2,Set(EXT_CALLED=${ARG1})
exten =
s,3,GotoIf($[${EXT_CALLED}=${EXT_STUDIO}]?s-STUDIO-${DIALSTATUS},1:s-REST-${DIALSTATUS},1)
exten = s-STUDIO-NOANSWER,1,Voicemail(u2)
exten = s-STUDIO-NOANSWER,2,Hangup()
exten = s-STUDIO-BUSY,1,Voicemail(b2)
exten = s-STUDIO-BUSY,2,Hangup()
exten = _s-STUDIO-.,1,Hangup()
exten = s-REST-NOANSWER,1,Voicemail(u1)
exten = s-REST-NOANSWER,2,Hangup()
exten = s-REST-BUSY,1,Voicemail(b1)
exten = s-REST-BUSY,2,Hangup()
exten = _s-REST-.,1,Hangup()
Is there any other way to obtain the same result but being easier to
configure??
Thanks in advance.
Best regards,
Jonathan GF
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--
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI SOLVED!!
Hi list, After talking with Digium, they shipped to me a TE120P, this card with his modern chipset has solved the noise issue. I'm very happy with Digium support, specially with Rod and Russell. This issue shows me that behind Digium Inc there are people who helps their customers with a great support. Thanks to all, Marc Matthew Fredrickson wrote: Arthur Miller wrote: The Digium cards are known to steal IRQ's. The Sangoma cards do not Not to appear defensive, but that is a technically inaccurate and also technically ambiguous statement. To correct it, there used to be a potential problem related to using the TE2xxP/TE4xxP cards relating to IRQ sharing which was fixed by a driver update. That is now resolved, and there shouldn't be any further issues. A considerable portion of the IRQ problems are an urban legend, a sort of scapegoat to point at. However, I would like to say that if anyone *does* have any problems relating to this, Digium and I personally are *very* interested in correcting them. We want to make sure that you trust our products, and want to stand behind our ability to support that. We have had some growing pains along the way, but we are *very* interested in making sure our hardware works to your and our other customers' satisfaction, and certainly stands up for itself in the face of competition. The Asterisk community is very important to us, and your perception of our products is crucial to our ability to afford to better support you and also forward the development of Asterisk. If you do have a problem, please contact technical support so that it can be fixed as soon as possible. ___ --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] Dificult macro, please advise
Atis, thanks for the quick post. I tried, probably wrong, to make a simple macro for all local switching, but i realized it became hard to mantain and can divert to errors in the future. I think i will go towards your proposal. Thanks for the input :) Jonathan GF On 9/3/07, Atis [EMAIL PROTECTED] wrote: On 9/3/07, Jonathan GF [EMAIL PROTECTED] wrote: Hi, BRIEF RESUME: Is there any other way to obtain the same result but being easier to configure?? Thanks! Why don't you use asterisk extension masks? ___ --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 with app_RPT question
Dear All, I am not sure if this is the right place to ask my question but I can't find a newsgroup or support for this app_RPT concept so I hope if some one in this community who have tried it out could help me out. I studied this application requirments and saw the hardware needed they describe a radio quad which uses RJ 45 but I can't see where the RJ goes in order to be able to communicate with the radio devices, I have absulotly no knowledge in two way radio devices so I hope some one could complete the full picture for me. And what I plan to use this feauture is to expand my * PBX ntwork to remote sites which have the radio quad and use it to talk to onsite engineers through the two way radio devices (such as motorola) using iaxrpt. -- Thx MAG ___ --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] How to handle + prefix
The Goto solutions worked fine for me, substitute a 00 for the +, then pass
onto our carriers who take it from there.
Thanks for all the feedback though. Most amusing.. :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse
Sent: 02 September 2007 17:45
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to handle + prefix
I had to do some work on the GUI for bandwidth, which sends
and matches + on every number.
To combat this, I just added one that says
exten = +extensionhere,1,Goto(${EXTEN:1})
exten = extensionhere,1,Noop(sweet.)
-bk
- Original Message -
From: Steve Murphy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, September 1, 2007 5:49:17 PM (GMT-0800) America/Los_Angeles
Subject: Re: [asterisk-users] How to handle + prefix
On Thu, 2007-08-30 at 10:17 -0500, Brian West wrote:
On Aug 30, 2007, at 10:11 AM, Jared Smith wrote:
On Thu, 2007-08-30 at 15:42 +0100, Adrian Marsh wrote:
Is there a way of using variables within the dialplan, eg:
[globals]
SOMEVAR=0179344
[local]
exten = _${SOMEVAR}.,1,NoOp(Dialled own number)
No, unfortunately you can't use variables as part of the extension
name
or pattern match.
Since when? I knew you couldn't use them for pattern matches but in
1.2 you could at one point I tested this personally.
Brian's right. But, the variable has to be a global, and is evaluated at
the time the extensions.conf file is read in; so, really, it's a
constant, and isn't evaluated at all at dial time. So, if you do one of
those dialplan show things, you'll see that the variable has been
substituted. Just had to fix a bug where AEL didn't provide the same
service.
murf
--
Steve Murphy
Software Developer
Digium
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[asterisk-users] Digium 4FXO PCI card or Grandstream 4FXO Gateway?
If you had the possibility to choose one, which will it be? I have been told that the PCI card is better, but the Gateway is easier to setup? Thanks in advance Alejandro Lengua ___ --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] ADIT 600 CMG = Asterisk question
I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 600 and route inbound calls towards asterisk. Can I have more than one CMG in a single chassis? Or maybe you know of a better way to connect T1's to asterisk without zaptel cards using SIP Trunks? Thanks Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ --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] Append Extension number sounds to Voice Mail Message?
Could some provide me a sample code to append the extension number in voice to the beginning of a voice mail message wav file before or after the message is saved? The idea is if the voice mail message wav file arrives from several sources, the listener will hear the 4 digit extension inside the voice message when played. Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ --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] ADIT 600 CMG = Asterisk question
Are you talking about PRI's? The ADIT's can't handle termination of PRI's, only DI. I use them all the time to breakout FXS/FXO's for incoming and outgoing analog lines, but they have a tendency to introduce lots of echo.I've had to use HWEC every time I use the 600. -D From: [EMAIL PROTECTED] on behalf of Barton Fisher Sent: Mon 9/3/2007 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ADIT 600 CMG = Asterisk question I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 600 and route inbound calls towards asterisk. Can I have more than one CMG in a single chassis? Or maybe you know of a better way to connect T1's to asterisk without zaptel cards using SIP Trunks? Thanks Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com winmail.dat___ --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] Manager Originate without phone off hook?
I'm trying to keep the DND status of my Snom phones and the astdb in
line but I'm stuck on integrating my gui DND button which talks to *
using the manager interface (actually it uses Astmanproxy as the gui
host is on a different network to asterisk and can't see the Snom's
across the network).
All's working fine in my Dialplan; when someone dials the code for
DND-on or DND-off I can do:
exten = *08,n,System(wget -qb -O /dev/null -o /dev/null
http://admin:[EMAIL PROTECTED]/dummy.htm?settings=savednd_mode=off)
exten = *08,n,Set(DB(DND/SIP/${MYSNOM})=0)
which turns the DND indicator on the phone off or on in line with the
database record. That's Great.
However, I'm completely flummoxed on getting a GUI DND button to work
sensibly via the Manager interface.
I could use 'Originate' to make the phone dial '*08' but that forces the
user to pickup the phone when they click the GUI DND button. Not Good :-(
So... can anyone suggest how I can use the Manager interface to set an
astdb record and send a request to the Snom to turn its DND indicator on
or off at the same time?
Thanks in advance.
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] ADIT 600 CMG = Asterisk question
Actually, these are old D4 SF (non-PRI) circuits - Could your echo be caused by FXO/FXS termination? I wonder if CMG would suffer as much as I believe it would stay 4 wire towards asterisk ? Bart Darren Wright wrote: Are you talking about PRI's? The ADIT's can't handle termination of PRI's, only DI. I use them all the time to breakout FXS/FXO's for incoming and outgoing analog lines, but they have a tendency to introduce lots of echo.I've had to use HWEC every time I use the 600. -D From: [EMAIL PROTECTED] on behalf of Barton Fisher Sent: Mon 9/3/2007 11:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ADIT 600 CMG = Asterisk question I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 600 and route inbound calls towards asterisk. Can I have more than one CMG in a single chassis? Or maybe you know of a better way to connect T1's to asterisk without zaptel cards using SIP Trunks? Thanks Bart -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com __ NOD32 2500 (20070903) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com begin:vcard fn:Barton Fisher n:Fisher;Barton org:Innovative Communications adr:;;7439 La Palma Ave # 255;Buena Park;CA;90620;USA email;internet:[EMAIL PROTECTED] tel;work:714-228-5410 url:http://icpage.com version:2.1 end:vcard ___ --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] Manager Originate without phone off hook?
May be I am missing something, but, manager command DBPut should do
the trick of putting the DB value. And, since you are already using
the manager interface, you are using PHP or PERL to connect to the
Database, why not wait for the DBPut command response and from the
script execute wget??
Regards
On 9/3/07, Russell Brown [EMAIL PROTECTED] wrote:
I'm trying to keep the DND status of my Snom phones and the astdb in
line but I'm stuck on integrating my gui DND button which talks to *
using the manager interface (actually it uses Astmanproxy as the gui
host is on a different network to asterisk and can't see the Snom's
across the network).
All's working fine in my Dialplan; when someone dials the code for
DND-on or DND-off I can do:
exten = *08,n,System(wget -qb -O /dev/null -o /dev/null
http://admin:[EMAIL PROTECTED]/dummy.htm?settings=savednd_mode=off)
exten = *08,n,Set(DB(DND/SIP/${MYSNOM})=0)
which turns the DND indicator on the phone off or on in line with the
database record. That's Great.
However, I'm completely flummoxed on getting a GUI DND button to work
sensibly via the Manager interface.
I could use 'Originate' to make the phone dial '*08' but that forces the
user to pickup the phone when they click the GUI DND button. Not Good :-(
So... can anyone suggest how I can use the Manager interface to set an
astdb record and send a request to the Snom to turn its DND indicator on
or off at the same time?
Thanks in advance.
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Within C++, there is a much smaller and cleaner language struggling
to get out.
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Re: [asterisk-users] Asterisk with app_RPT question
Hi Mohamed, See at [EMAIL PROTECTED] and http://app-rpt.qrvc.com Best Regards, F6HQZ Francois BERGERET France___ --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] Cepstral's Allison is having trouble speaking clearly
Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? TIA, Todd Reese ___ --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] Cepstral's Allison is having trouble speaking clearly
Todd Reese wrote: Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? Make sure that you have RTP Packet Size changed to 0.02 from the default 0.03 in the Sipura SIP tab. This is known to cause jitter with Asterisk. HTH, Vahan ___ --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] Cepstral's Allison is having trouble speaking clearly
Try setting the RTP packets to 0.020 instead of 0.030 which is the default on the SPA's /b On Sep 3, 2007, at 5:00 PM, Todd Reese wrote: Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? TIA, Todd Reese ___ --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] Cepstral's Allison is having troublespeaking clearly
OK, I just reset the RTP packets to .020 as you have suggested. I can tell a little difference but the problem is still there. TIA, Todd - Original Message - From: Brian West [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 03, 2007 6:10 PM Subject: Re: [asterisk-users] Cepstral's Allison is having troublespeaking clearly Try setting the RTP packets to 0.020 instead of 0.030 which is the default on the SPA's /b On Sep 3, 2007, at 5:00 PM, Todd Reese wrote: Hi all, I have just install and licensed Cepstral's Allison08kHz on my Asterisk 1.4.11 system. I can call the Allison's extension from my Grandstream IP Phone and she's clear as a bell, but when a call to her extension traverses through one of the Linksys/Sipura 3102 or 2002, she's got the jitters bad. The SPA-202 has only an extension phone on it and the SPA-3102 is my FXO from my Vonage Motorola box. Any clues where to start looking to clear this up? TIA, Todd Reese ___ --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] FW: Account Registration Failed
Hi, I am trying to run an Asterisk (1.4.11) server on Linux Suse. The server is behind NAT. I am testing with SIP client that developed from PJSIP running on Pocket PC Windows Mobile 5.0 . The client is also behind another NAT. STUN server is implement in SIP client. When a register Message send from client to server, Asterisk receive it and reply with 100 trying msg. However, there is no reply on 200 OK from server, as it course my SIP client registration failure. 1. On the other hand, if i tested with OPENSER SIP server, registration is fine. Important details are below: sip.conf [global] nat=yes canreinvite=no localnet=192.168.1.46 externip=60.xx.xx.xx.xx [8000] type=friend secret=8000 nat=yes host=dynamic canreinvite=no How can i solve it? p/s : Network traffic capture in Ethereal are attached. ~ cobra client.cap - capture at client side ~ cobra server - capture at Asterisk server Thanks Regards kum Live Search: Better results, fast Try it now! _ Personalize your Live.com homepage with the news, weather, and photos you care about. http://www.live.com/getstarted.aspx?icid=T001MSN30A0701 cobra client.cap Description: Binary data cobra server.cap Description: Binary data ___ --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] FW: Account Registration Failed
Localnet is wrong... try localnet=192.168.1.0/24 /b On Sep 3, 2007, at 9:13 PM, neoh kumyee wrote: Hi, I am trying to run an Asterisk (1.4.11) server on Linux Suse. The server is behind NAT. I am testing with SIP client that developed from PJSIP running on Pocket PC Windows Mobile 5.0 . The client is also behind another NAT. STUN server is implement in SIP client. When a register Message send from client to server, Asterisk receive it and reply with 100 trying msg. However, there is no reply on 200 OK from server, as it course my SIP client registration failure. 1. On the other hand, if i tested with OPENSER SIP server, registration is fine. Important details are below: sip.conf [global] nat=yes canreinvite=no localnet=192.168.1.46 externip=60.xx.xx.xx.xx [8000] type=friend secret=8000 nat=yes host=dynamic canreinvite=no How can i solve it? p/s : Network traffic capture in Ethereal are attached. ~ cobra client.cap - capture at client side ~ cobra server - capture at Asterisk server Thanks Regards kum Live Search: Better results, fast Try it now! Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it! cobra client.cap cobra server.cap ___ --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] Testing Framework
matt, are you looking for unit testing of the * components or systems testing, testing the finished product? or both? I think you are onto something here... I hope it takes root. I would say put it in the addons. it would be Great if digium takes it up. it is a smart move for them to foster, cajole, nudge, and support it. call volume I would leave to others as different processors, O/S, builds, kernel versions, and configurations will have too many variables. I was playing with the idea of monitoring multiple * systems. perhaps we can start out with testing the components and then migrate the project (future) to one pbx monitor the other. we will need scripts to initiate some action, config to make some measurements, the scripts to gather the results into a nice neat little summary report. you will want to take the human aspect out of the picture as much as possible. for example: on pbx A * create a recording in multiple formats .gsm, .wav, etc. * initiate a script to generate 5,10, or 25 calls to pbx B and play the file on pbx B * pbx B gets the calls, records them, * copy the recordings from pbx A to pbx B (or have that already done) * have a wave analyzer compare the recordings to the original files (you know I won't be writing that program! :) * report on anomalies *call * *Technology * *recording delta * 1 Zap Provider 1 2% 2 VoIP Provider 2 5% 3 VoIP Provider 2 15% ... VoIP Provider 3 ... let me know what you think! daveC Matt Riddell wrote: Hash: SHA1 Hi, So, now that we've all complained about the state of testing of Open Source versions of Asterisk, lets do something about it. I propose we start with a list of things that we think should be tested in Asterisk, and means to test them. Maybe we could run certain tests based on the changes between minor versions? Anyway lets start. Call Volumes 1) Call volume up to x channels from SIP to SIP (i.e. sipp) 2) Call volume up to x channels from IAX2 to SIP 3) Call volume up to x channels from IAX2 to IAX2 Application testing 4) Connect x calls between techs to Meetme (leave running for 1 hour) 5) Connect x concurrent calls to VoiceMail Call Centre Testing 6) Send x calls to a queue with no agents in it, leave them holding for x minutes 7) Run x calls against AMD connected to recorded known good files Recording 8) Run x calls recording simultaneously from an automatically generated call, play ulaw/alaw - compare outputs. You get the idea. If people can add to this list, I can start making a few scripts and programs that will test them (as I'm sure others can). If we end up with a complete list, I'm sure some of our individual QA departments can take the responsibility for certain items. The call volume ones are obviously going to either need a live person to dial in at volume and check everything is ok, or a recording which can later be checked. I'm of the opinion that the majority of tests should test individual components, but that we should also form some Application Type frameworks so that we can test integration between Asterisk apps. Any takers? Add to the list? If there is something you believe is mission critical to your business, write up a test case for it, and we'll all try to code something that can run automatically to test it. If we try and keep to ANSI C for the testing apps, Digium should be able to run them on their multi platform machines as well. Should these tests be added to Asterisk-Addons or maintained outside of the tree? Anyway, what do you think? Feasible? I already have a few tests here and I'm sure others have a few too. Lets put them all together and get a framework going. - -- 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://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG1yKhDQNt8rg0Kp4RAv5UAJ48tW28T5lWCQIPTwVimyvlhEPJowCgpnE6 OF3L2M/6Hc+YBNL1NFx6dzA= =OXNn -END PGP 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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing
Re: [asterisk-users] E1 to Ethernet Bridge
Please can you go into further detail. Thanks.. From: Jared Smith [EMAIL PROTECTED] Sent: Friday, 31 August, 2007 2:25:23 PM On Fri, 2007-08-31 at 12:55 +, Arinze Izukanne wrote: I am trying to Bridge 2 E1 interfaces over a long distance link It's really not too difficult... it ends up looking like this: E1 - Asterisk A - IAX2 - Asterisk B - E1 Asterisk box A is configured to take calls from the E1, and send the calls over IAX2 to box B. Box B takes calls in over IAX2, and sends them out its E1. (The reverse can also be setup as well, so that calls coming into box B over its E1 can be forwarded over IAX2 to box A, and so forth.) Let me know if you need me to go into further detail. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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 ___ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/ ___ --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] E1 to Ethernet Bridge
Can you show me a sample fo config? The link schematic should look like this: E1 == TDMoE==E1. From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, 31 August, 2007 7:03:03 PM Subject: Re: [asterisk-users] E1 to Ethernet Bridge On Fri, Aug 31, 2007 at 12:55:24PM +, Arinze Izukanne wrote: Hello, I am trying to Bridge 2 E1 interfaces over a long distance link exactly the same way Redfone does. How can asterisk be configured to do that? If I understand correctly, yyou should configure those two E1 interfaces in Zaptel alone (and don't run Asterisk) and then expose them as dynamic spans (ztd-eth ). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html___ --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] FW: Account Registration Failed
Hi , I have change the localnet to = 192.168.1.0/24, but still can't receive any 200 OK reply message from asterisk. I have check with Ethereal, no 200 OK send out from asterisk... why? ? if i use other AGEPhone, registration is success... Any solution to solve it?? thanks From: [EMAIL PROTECTED] Date: Mon, 3 Sep 2007 21:29:29 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FW: Account Registration Failed Localnet is wrong... try localnet=192.168.1.0/24/b On Sep 3, 2007, at 9:13 PM, neoh kumyee wrote: Hi, I am trying to run an Asterisk (1.4.11) server on Linux Suse. The server is behind NAT. I am testing with SIP client that developed from PJSIP running on Pocket PC Windows Mobile 5.0 . The client is also behind another NAT. STUN server is implement in SIP client. When a register Message send from client to server, Asterisk receive it and reply with 100 trying msg. However, there is no reply on 200 OK from server, as it course my SIP client registration failure. 1. On the other hand, if i tested with OPENSER SIP server, registration is fine. Important details are below: sip.conf [global] nat=yes canreinvite=no localnet=192.168.1.46 externip=60.xx.xx.xx.xx [8000] type=friend secret=8000 nat=yes host=dynamic canreinvite=no How can i solve it? p/s : Network traffic capture in Ethereal are attached. ~ cobra client.cap - capture at client side ~ cobra server - capture at Asterisk server Thanks Regards kum Live Search: Better results, fast Try it now! Call and stay connected with your friends and family for free. Seen and be heard with high-definition video calls on Windows Live Messenger. Try it!cobra client.capcobra server.cap___--Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Personalize your Live.com homepage with the news, weather, and photos you care about. http://www.live.com/getstarted.aspx?icid=T001MSN30A0701___ --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] VSP authentication to incorrect context
All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch, have GoTalk, digest has 09xx ;GoTalk Outbound [GoTalk] username=09xx fromuser=09xx fromdomain=sip.gotalk.com type=peer secret= qualify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xx] username=09xx type=user secret= fromuser=09xx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xx:[EMAIL PROTECTED]/09xx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] , www.intergraph.com.au http://www.intergraph.com.au ___ --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] unsuscribe
please unsubscribe Moshe Wahrhaftig IT Manager Talk'n'Save Israel: 02-655-0313 Cell: 052-2771738 USA: 516-204- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Rodriguez Sent: Monday, September 03, 2007 10:51 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Rechazo de llamada en triangulacion deasterisk. Gracias Alex. Lo pondré el Lunes y te tire como van las cosas. Guillermo El Jueves, 30 de Agosto de 2007 17:52, Alex Balashov escribió: Guillermo, Me parece que la cosa aqui es que el nombre del usuario debe ser el mismo en el URI del fuente que en el el proceso de autentificacion. Traiga poner username= en la configuracion asi: On Thu, 30 Aug 2007, Guillermo Rodriguez wrote: [pbx1] name=test1 callerid=200 host=dynamic nat = yes type friend secret= test1 username=... Y diganos lo que pasa. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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
Re: [asterisk-users] VSP authentication to incorrect context
This links seems to show that insecure=very might need to be set later, PaulH http://forums.whirlpool.net.au/forum-replies-archive.cfm/359239.html On Tue, 2007-09-04 at 13:50 +1000, Klaverstyn, David C wrote: All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch, have GoTalk, digest has 09xx ;GoTalk Outbound [GoTalk] username=09xx fromuser=09xx fromdomain=sip.gotalk.com type=peer secret= qualify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xx] username=09xx type=user secret= fromuser=09xx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xx:[EMAIL PROTECTED]/09xx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 [EMAIL PROTECTED], www.intergraph.com.au ___ --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] VSP authentication to incorrect context
Many thanks, that did the trick. I actually read that page previously. I'm Not sure why it did not work or why I did not try entering that line previously. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Tuesday, 4 September 2007 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VSP authentication to incorrect context This links seems to show that insecure=very might need to be set later, PaulH http://forums.whirlpool.net.au/forum-replies-archive.cfm/359239.html On Tue, 2007-09-04 at 13:50 +1000, Klaverstyn, David C wrote: All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch, have GoTalk, digest has 09xx ;GoTalk Outbound [GoTalk] username=09xx fromuser=09xx fromdomain=sip.gotalk.com type=peer secret= qualify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xx] username=09xx type=user secret= fromuser=09xx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xx:[EMAIL PROTECTED]/09xx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 [EMAIL PROTECTED], www.intergraph.com.au ___ --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
