[asterisk-users] How to originate a call from console CLI ?
Hi, I would like to originate my first call from CLI. As I'm new to this, I'm wondering if it's possible. When I type "originate" from CLI, I've got this : " There are two ways to use this command. A call can be originated between a channel and a specific application, or between a channel and an extension in the dialplan. This is similar to call files or the manager originate action. Calls originated with this command are given a timeout of 30 seconds. Usage1: originate application [appdata] This will originate a call between the specified channel tech/data and the given application. Arguments to the application are optional. If the given arguments to the application include spaces, all of the arguments to the application need to be placed in quotation marks. Usage2: originate extension [EMAIL PROTECTED] This will originate a call between the specified channel tech/data and the given extension. If no context is specified, the 'default' context will be used. If no extension is given, the 's' extension will be used." I would like for example to call 0123456789 number from SIP/7530 extension. My asterisk server is set to use "local" context for outgoing calls. My first idea was to type this : originate SIP 7530 [EMAIL PROTECTED] But it fails : it keeps displaying " There are two ways ..." and nothing else seem to occur. Can anyone help ? 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
Re: [asterisk-users] Asterisk on Pcengines Alix board
> Thanks for the tip. It seems like they no longer manufacture them: > > http://www.neoware.com/products/hardware/ No, but the Neoware e140 has a PCI expansion slot, is expandable to 1GB RAM, and still has room inside the case for a hard drive. It is available without Win XPe starting at $339 new. The prices on these are coming down. > - Fan-less, compact motherboard While some of these thin clients have fans, many of them only run the fan when set horizontally and thus don't have the advantage of heat induced air flow. > - hard-disk (so I don't have to tweek Linux making too many writes and > wear down the CF card) The newer CF cards are making this nearly a mute point. Seems like I provide updated software often enough that I never have CF cards wear out. We format a new image for the customer and send out a new card. They take a couple of minutes to power down the system, swap out the card and boot up on the new load. When done they return the old card to us for "recycling" and get a $10 credit. > - a PCI card installed at an 90° angle (I prefer to use a PCI card > instead of an external FXO gateway) I'm not sure I understand the need for the PCI card to be perpendicular to the board. I prefer the flatter box since they mount to a wall well and provide a nice compact installation. John ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newb Question
you can try Cain & Abel ( to route calls) and Wireshark to record all the calls. On 11/29/07, Adam Moffett <[EMAIL PROTECTED]> wrote: > > I'm pretty sure asterisk won't do that without modification. You'll > need to do packet sniffing and decode the datathere may be products > that do this, but asterisk is not it. > > And we're assuming the calls are unencrypted? > > I inherited an office with phones that are hosted off-site. Everything > > is skinny and G729. I see that the FreeBSD asterisk port comes with a > > G729 codec. > > > > I want to record everything. If I use port mirroring on my switch, is > > it possible to configure asterisk to record and assemble packets that > > it doesn't otherwise route? Is it insane to user asterisk for this > > purpose? Advice or a link to a howto would be greatly appreciated. > > > > > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct syntax for IF()?
On Mon, 2007-11-26 at 21:23 -0500, Adam Moffett wrote:
> A simpler example reveals the problem:
>
> exten => 188,1,Noop(${STAT(e,/bin/ls)})
> exten => 188,2,Noop(${STAT(e,/not/there)})
>
> Try that and you'll find that STAT(e,/whatever) returns 1 if the file is
> found and NOTHING if the file is not found.
Sounds like a perfect application for the ISNULL dialplan function. Of
course, that adds a whole new set of curly braces and parentheses to
watch out for.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Newb Question
I'm pretty sure asterisk won't do that without modification. You'll need to do packet sniffing and decode the datathere may be products that do this, but asterisk is not it. And we're assuming the calls are unencrypted? > I inherited an office with phones that are hosted off-site. Everything > is skinny and G729. I see that the FreeBSD asterisk port comes with a > G729 codec. > > I want to record everything. If I use port mirroring on my switch, is > it possible to configure asterisk to record and assemble packets that > it doesn't otherwise route? Is it insane to user asterisk for this > purpose? Advice or a link to a howto would be greatly appreciated. > > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Parking/Pickup on a single button
At 03:10 PM 11/29/2007, you wrote: >Is it possible with asterisk to use a single button to park and >retrieve a call? I could do this with my Aastra 480i CT as the buttons can have different meaning for different states. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct syntax for IF()?
On Mon, 26 Nov 2007 21:23:59 -0500, Adam Moffett <[EMAIL PROTECTED]>
wrote:
>This method should work:
>
>${IF($["${STAT(e,/tmp/${CALLTIME}.wav)}" = "1"]?${CALLTIME}.wav:"")}
Yes indeed :-)
===
[internal]
exten => 888,1,Playback(leave_msg)
exten => 888,n,Set(CALLTIME=${STRFTIME(${EPOCH},,%d-%b-%Y-%Hh%M)})
exten => 888,n,Record(/tmp/${CALLTIME}.wav,3,30,s)
exten => 888,n,Hangup()
exten => h,1,NoOp(Let's dance)
exten => h,n,Set(WAV_FILE=${IF($["${STAT(e,/tmp/${CALLTIME}.wav)}" =
"1"]?${CALLTIME}.wav:"")})
exten => h,n,Verbose(WAV_FILE is ${WAV_FILE})
===
Thanks a lot for the help.
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Re: [asterisk-users] Newb Question
On Fri, 30 Nov 2007, ram wrote: > chan spy does the job i belive > > ram > > On Nov 30, 2007 7:37 AM, Jeff Adams <[EMAIL PROTECTED]> wrote: > >> I inherited an office with phones that are hosted off-site. Everything is >> skinny and G729. I see that the FreeBSD asterisk port comes with a G729 >> codec. >> I want to record everything. If I use port mirroring on my switch, is it >> possible to configure asterisk to record and assemble packets that it >> doesn't otherwise route? Is it insane to user asterisk for this purpose? >> Advice or a link to a howto would be greatly appreciated. Chanspy lets you listen to a channel. While it will record to a file, it would be a manual operation for every call. I suspect either you want to insert an Asterisk system in-between as a "tap" (requiring re-configuring your phones and your outside provider) or using a "voip sniffer" plugged into the management port of your Ethernet switch. Of course, I've done neither, so my advice is worth every penny you paid for it :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip 1.4.x DTMF detection not working
Hello
I have a setup where i have 2 asterisk servers connected over the public
internet with plenty of bandwidth, NAT on one side only. If i use IAX
between the two *'s dtmf is flawless. If I use SIP, DTMF detection is around
30% or less. I have an exten to dial into and check DTMF:
exten => NPANXX,1,Answer(); (actual number blanked for privacy)
exten => NPANXX,n,Read(userChoice|ogm/intro|4||1|4);
exten => NPANXX,n, SayDigits(${userChoice});
exten => NPANXX,n,Hangup();
When i dial in and use IAX between the servers i always get all 4 digits, If I
dial in using SIP between the two servers with dtmfmode=rfc2833 or
dtmfmode=inband I MIGHT get 1 or 2 digits. If i use dtmf=info and I dial
slowly I usually get 4 correct digits, but not consistently enough to call it
good, maybe 85%. If I dial 1 2 3 4 quickly I get 1122 or 1223 or the like.
I would like to use SIP as the voice quality "seems" to be better, matter of
opinion I am sure but...
Both Asterisk's are 1.4.x on SUSE 10.2 x86_64 kernel 2.6.18.2-34
AMD opteron Dual-Core AMD Opteron(tm) Processor 2212
and
Dual Core AMD Opteron(tm) Processor 180
2GIG memory
I have searched voip-info and google and didn't find anything that looked
relevant, maybe just my search words. I do seem to remember something on the
list about this a couple months ago but I can not find it or I am remembering
incorrectly.
Any suggestions will be greatly appreciated.
Thank You,
JohnM
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Re: [asterisk-users] Newb Question
chan spy does the job i belive ram On Nov 30, 2007 7:37 AM, Jeff Adams <[EMAIL PROTECTED]> wrote: > I inherited an office with phones that are hosted off-site. Everything is > skinny and G729. I see that the FreeBSD asterisk port comes with a G729 > codec. > I want to record everything. If I use port mirroring on my switch, is it > possible to configure asterisk to record and assemble packets that it > doesn't otherwise route? Is it insane to user asterisk for this purpose? > Advice or a link to a howto would be greatly appreciated. > > ___ > --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] Newb Question
I inherited an office with phones that are hosted off-site. Everything is skinny and G729. I see that the FreeBSD asterisk port comes with a G729 codec. I want to record everything. If I use port mirroring on my switch, is it possible to configure asterisk to record and assemble packets that it doesn't otherwise route? Is it insane to user asterisk for this purpose? Advice or a link to a howto would be greatly appreciated. ___ --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] Adding new recorded phrases to the release
On Thursday 29 November 2007 13:29:42 Philip Prindeville wrote: > This might be a frequently asked question, but how do new sounds get > added to the release? Patches that use new sounds have to be added to the bugtracker, and Digium pays for sounds to be recorded in the 3 languages that we distribute for: English, Spanish, and French. -- Tilghman ___ --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] AST-2007-026 - SQL Injection issue in cdr_pgsql
Asterisk Project Security Advisory - AST-2007-026 ++ | Product| Asterisk| |--+-| | Summary| SQL Injection issue in cdr_pgsql| |--+-| | Nature of Advisory | SQL Injection | |--+-| |Susceptibility| Remote Authenticated Sessions | |--+-| | Severity | Moderate| |--+-| |Exploits Known| No | |--+-| | Reported On | November 29, 2007 | |--+-| | Reported By | Tilghman Lesher | |--+-| | Posted On | November 29, 2007 | |--+-| | Last Updated On| November 29, 2007 | |--+-| | Advisory Contact | Tilghman Lesher | |--+-| | CVE Name | CVE-2007-6170 | ++ ++ | Description | Input buffers were not properly escaped when providing | | | the ANI and DNIS strings to the Call Detail Record | | | Postgres logging engine. An attacker could potentially | | | compromise the administrative database containing users' | | | usernames and passwords used for SIP authentication, | | | among other things. | | | | | | This module is not active by default and must be | | | configured for use by the administrator. Default | | | installations of Asterisk are not affected. | ++ ++ | Workaround | Convert your installation to use cdr_odbc with the| || PgsqlODBC driver. This module provides similar| || functionality but is not vulnerable. | ++ ++ |Resolution| Upgrade to Asterisk release 1.4.15 or higher. | ++ ++ | Affected Versions| || |Product| Release | | | | Series| | |---+-+--| | Asterisk Open Source |1.0.x| All versions | |---+-+--| | Asterisk Open Source |1.2.x| 1.2.24 and previous | |---+-+--| | Asterisk Open Source |1.4.x| 1.4.14 and previous | |---+-+--| | Asterisk Business Edition |A.x.x| All versions | |---+-+--| | Asterisk Business Edition |B.x.x| B.2.3.3 and previous | |---+-+--| | Asterisk Business Edition |C.x.x| C.1.0-beta5 and previous | |---+-+--| | AsteriskNOW | pre-release | None
[asterisk-users] Call Parking/Pickup on a single button
Is it possible with asterisk to use a single button to park and retrieve a call? e.g. Button is labelled Park 701 - If it is not in use, park the current call to 701 - If it is in use (the associated LED will be lit), pickup the call at 701 (putting the current call [if any] on hold). A Polycom IP600 phone would have two or three of these keys (701, 702, 703). Any suggestions appreciated! Alvin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
On Thu, 29 Nov 2007 00:06:38 -0600, "John Faubion" <[EMAIL PROTECTED]> wrote: >Many of the thin clients fit the bill nicely. I've been using MaxSpeed >MaxTerm clients lately. Thanks for the tip. It seems like they no longer manufacture them: http://www.neoware.com/products/hardware/ I'll look in the archives of the list to see what people have posted about this. I'd really like to come up with a compact solution, ideally: - Fan-less, compact motherboard - hard-disk (so I don't have to tweek Linux making too many writes and wear down the CF card) - a PCI card installed at an 90° angle (I prefer to use a PCI card instead of an external FXO gateway) I bought a Via motherboard in a compact case once, but 500E is a bit pricey. once 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
[asterisk-users] Adding new recorded phrases to the release
This might be a frequently asked question, but how do new sounds get added to the release? I was trying to do a "popcorn" extension on my phone (that gives the date and time... maybe even getting fancy and adjusting for the caller's timezone based on country code or area code)... but didn't have the word "local" or phrase "local time" in the lexicon. Now if I could just figure out how to grab time current time as UNIX seconds... add a small delay to it (like 5, the time it takes to sound out the time), and then wait for that time... then play a sychronizing tone... then I'll be all done: [popcorn] exten => s,1,Answer() exten => s,n,SayUnixTime(,Zulu,HNS) exten => s,n,SayPhonetic(z) exten => s,n,SayUnixTime(,,HNS) exten => s,n,Playback(vm-localtime) exten => s,n,Return() ___ --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] Using existing extensions.conf macros, and co-habitation
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no "default" context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the "default" context:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten => a,1,VoicemailMain(${ARG1})
The issue is that I have, per "virtual pbx" (i.e. home or business), two
contexts
that these get used from. The "internal-xyzzy" and "incoming-xyzzy" contexts
(one
for each pbx, ie. "xyzzy" is "home" or else it's "office").
I was wondering if there wasn't a more flexible solution to this issue, than
hard-coding a "Goto(default,s,1)" into them (I have no default context, because
it
would be meaningless).
Perhaps using "Gosub" and "Return". Or do I need to hack the macro, and pass
in a
3rd argument (bletch)?
Is this doable?
Thanks,
-Philip
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[asterisk-users] AST-2007-025 - SQL Injection issue in res_config_pgsql
Asterisk Project Security Advisory - AST-2007-025 ++ | Product| Asterisk| |--+-| | Summary| SQL Injection issue in res_config_pgsql | |--+-| | Nature of Advisory | SQL Injection | |--+-| |Susceptibility| Remote Unauthenticated Sessions | |--+-| | Severity | Moderate| |--+-| |Exploits Known| No | |--+-| | Reported On | November 29, 2007 | |--+-| | Reported By | P. Chisteas | |--+-| | Posted On | November 29, 2007 | |--+-| | Last Updated On| November 29, 2007 | |--+-| | Advisory Contact | Tilghman Lesher | |--+-| | CVE Name | | ++ ++ | Description | Input buffers were not properly escaped when providing | | | lookup data to the Postgres Realtime Engine. An attacker | | | could potentially compromise the administrative database | | | containing users' usernames and passwords used for SIP | | | authentication, among other things. | | | | | | This module is not active by default and must be | | | configured for use by the administrator. Default | | | installations of Asterisk are not affected. | ++ ++ | Workaround | Convert your installation to use res_config_odbc with the | || PgsqlODBC driver. This module provides similar| || functionality but is not vulnerable. | ++ ++ |Resolution| Upgrade to Asterisk release 1.4.15 or higher. | ++ ++ | Affected Versions| || | Product| Release | | | | Series| | |--+-+---| | Asterisk Open Source |1.0.x| None | |--+-+---| | Asterisk Open Source |1.2.x| None | |--+-+---| | Asterisk Open Source |1.4.x| 1.4.14 and previous | | | | versions | |--+-+---| | Asterisk Business Edition |A.x.x| None | |--+-+---| | Asterisk Business Edition |B.x.x| None | |--+-+---| | AsteriskNOW | pre-release | None | |--+-+---| | Asterisk Appliance Developer |0.x.x| None
[asterisk-users] Asterisk 1.4.15 and 1.2.25 Released
The Asterisk.org development team has released Asterisk versions 1.4.15 and 1.2.25. These releases contain two fixes for security issues. http://downloads.digium.com/pub/asa/AST-2007-025.pdf * This is a SQL injection vulnerability in the res_config_pgsql module. Default installations of Asterisk are not affected. However, any system using the Postgres Realtime Engine may be remotely exploitable. This issue only affects Asterisk 1.4, as this module was not in Asterisk 1.2. http://downloads.digium.com/pub/asa/AST-2007-026.pdf * This is another SQL injection vulnerability. The input for the ANI and DNIS fields were not properly escaped. Default installations of Asterisk are not vulnerable. However, systems that use the Postgres CDR logging module may be remotely exploitable. This issue affects both Asterisk 1.2 and 1.4. Both releases are available on http://downloads.digium.com. Thank you very much for your support! ___ --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] Transfering IAX context
[EMAIL PROTECTED] wrote: > Hello Everyone, > > I have a 2 Asterisk Servers, one in US and another in India. > Once someone from US calls, call hit US server and then is forwarded to India > which then is answered by someone. > i.e. > Caller --> US Asterisk Server --> India Asterisk Server --> Employee(India) > > > The Employee in India decides that the call was for Employee in US, so he > transfer the call to the employee in US. > i.e. > Caller --> US Asterisk Server --> India Asterisk Server --> Employee(India) > --> India Asterisk Server --> US Asterisk Server --> Employee (US) > OR > Caller --> US Asterisk Server --> India Asterisk Server --> US Asterisk > Server --> Employee (US) > > (Not sure which explanation is correct as per asterisk working, but hopefully > should be the second.) > > > The way this type of communication traverse is that the call has to come to > India and the reverted back to US. > Is their a way that when a Employee in India transfers back the call to US > Asterisk Server, the Indian server should completely removed from the > picture. This would save our Bandwidth utilization. > i.e. flow becomes :: > Caller --> US Asterisk Server --> Employee (US) > Usually, yes, the call traffice will get rerouted from one server to another. If you watch your console on the US server you should even be able to see this happen. This can even work in a 3 server setup where server A calls B and then B transfers to C. Server A will try to contact server C directly and if it can, drop the communications between A and B and B and C. Not sure if it works the same way with SIP. -Dave ___ --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] Transfering IAX context
Hello Everyone, I have a 2 Asterisk Servers, one in US and another in India. Once someone from US calls, call hit US server and then is forwarded to India which then is answered by someone. i.e. Caller --> US Asterisk Server --> India Asterisk Server --> Employee(India) The Employee in India decides that the call was for Employee in US, so he transfer the call to the employee in US. i.e. Caller --> US Asterisk Server --> India Asterisk Server --> Employee(India) --> India Asterisk Server --> US Asterisk Server --> Employee (US) OR Caller --> US Asterisk Server --> India Asterisk Server --> US Asterisk Server --> Employee (US) (Not sure which explanation is correct as per asterisk working, but hopefully should be the second.) The way this type of communication traverse is that the call has to come to India and the reverted back to US. Is their a way that when a Employee in India transfers back the call to US Asterisk Server, the Indian server should completely removed from the picture. This would save our Bandwidth utilization. i.e. flow becomes :: Caller --> US Asterisk Server --> Employee (US) Regards, Sanjay. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP detects loop when forwarding to voicemail
Thank you!
It would be greate to have these feature set as a parameter in sip.conf
cheers
tomasz
On Nov 28, 2007 2:38 PM, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
> Tomasz Zieleniewski wrote:
> > How does asterisk detect the loop.
> > What are the criteria here.
> > What do I need to change in the SIP message so
> > that asterisk will not consider it looped??
>
> > On Nov 23, 2007 4:03 PM, Tomasz Zieleniewski <[EMAIL PROTECTED]>
> > wrote:
> >
> >> hi,
> >>
> >> I use asterisk as a gateway which forwards external calls from pstn to
> >> my internal sip network.
> >> all sip signaling is passed to sip proxy.
> >> I also use asterisk as a voicemail server.
> >> everything works well when calls are passed to asterisk from local
> >> network.
> >> but when calls are forwarded from asterisk to sip proxy and then sip
> >> proxy decides to pass it back to asterisk
> >> waorking as a voicemail server
> >> asterisk complains about the loop and returns 482 response.
> >> Can it be somehow reconfigured??
>
> See
> http://bugs.digium.com/view.php?id=7403
> and look for this code in chan_sip.c:
> ---cut---
>/* Check if this is a loop */
>if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner &&
> (p->owner->_state != AST_STATE_UP)) {
>/* This is a call to ourself. Send ourselves an error code
> and stop
>processing immediately, as SIP really has no good mechanism
> for
>being able to call yourself */
>/* If pedantic is on, we need to check the tags. If they're
> different, this is
>in fact a forked call through a SIP proxy somewhere. */
>transmit_response(p, "482 Loop Detected", req);
>sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
>return 0;
>}
> ---cut---
>
> There's no way to configure the loop detection but you could
> remove the code to disable loop detection.
>
> Grüße,
> Philipp Kempgen
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
>Let's use IT to solve problems and not to create new ones.
> Asterisk? -> http://www.das-asterisk-buch.de
>
> Geschäftsführer: Stefan Wintermeyer
> Handelsregister: Neuwied B 14998
>
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Re: [asterisk-users] Problems with Asterisk 1.4.14 and Queue app
You are right!
Here there is the backtrace
(gdb) bt
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
"default", interpclass=0x0) at res_musiconhold.c:646
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 , interpclass=0x2e0 ) at channel.c:4609
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 "my-queue", exten=0x821d090 "80", priority=8, label=0x0,
callerid=0x819dc90 "490814", action=E_MATCHMORE) at pbx.c:532
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
(gdb) bt full
#0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
No symbol table info available.
#1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
"default", interpclass=0x0) at res_musiconhold.c:646
mohclass = (struct mohclass *) 0x2e0
#2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 , interpclass=0x2e0 ) at channel.c:4609
No locals.
#3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
app_queue.c:3600
makeannouncement = 0
res = 136429000
ringing = 0
lu = (struct ast_module_user *) 0x8250230
user_priority = 0x821bdc8 "1196248345.116"
max_penalty_str = 0x821bdc8 "1196248345.116"
prio = 0
max_penalty = 0
reason = QUEUE_UNKNOWN
tries = 0
noption = 0
args = {argc = 5, argv = 0xb71a3908, queuename = 0xb71a3710
"my-queue", options = 0xb71a371b "t", url = 0xb71a371d "",
announceoverride = 0xb71a371e "", queuetimeoutstr = 0xb71a371f "300", agi
= 0x0}
qe = {parent = 0x8227198, moh = "default", '\0' ,
announce = '\0' , context = '\0' ,
digits = '\0' , valid_digits = 0, pos = 1, prio = 0,
last_pos_said = 0, last_periodic_announce_time = 1196248351,
last_periodic_announce_sound = 0, last_pos = 0, opos = 1, handled = 0,
max_penalty = 0, start = 1196248351, expire = 1196248651, chan = 0x821cec0,
next = 0x0}
#4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
context=0x821d040 "my-queue", exten=0x821d090 "80", priority=8, label=0x0,
callerid=0x819dc90 "490814", action=E_MATCHMORE) at pbx.c:532
e = (struct ast_exten *) 0x8255fc0
res = 0
q = {incstack = {0x0 }, stacklen = 0, status = 5,
swo = 0x0, data = 0x0, foundcontext = 0x821d040 "my-queue"}
passdata = "my-queue|t|||300", '\0'
matching_action = 136488696
#5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
dst_exten = '\0' ,
"1¶Þ·\000\000\000\000\000\000\000\000\216\227ñ·", '\0' , "
Êé· Êé·D\236\032·ôßñ·
Êé·\f\000\000\000D\236\032·t\222ñ·0Êé·ô¯é·|\236\032·¥²Þ·
Êé·ôßñ·\200\026%\b¨x\024\b|\236\032·\215añ·\020\000\000\000\f\000\000\000pÍ%\b°Ò!\bÐL\"\b\000\000\000\000!\224ñ·\201
\006\b"
pos = 0
digit = 0
found = 1
res = 0
error = 0
#6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
No locals.
#7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
_buffer = {__routine = 0x8067ef0 , __arg =
0x1b8019, __canceltype = -1222992148, __prev = 0x0}
ret = (void *) 0x8224cd0
---Type to continue, or q to quit---
a = {start_routine = 0x80c78e0 , data = 0x821cec0,
name = 0x8224cd0 "pbx_thread", ' ' , "started at [ 2632]
pbx.c ast_pbx_start()"}
#8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9 0xb7e3e1ba in clone () from /lib/libc.so.6
No symbol table info available.
Thanks
On Nov 29, 2007 3:04 PM, Jared Smith <[EMAIL PROTECTED]> wrote:
> On Thu, 2007-11-29 at 14:28 -0300, equis software wrote:
> > I have problems with 1.4.14, it crash every few minutes.
> > The same configuration and machine in Asterisk 1.4.6 it doesn´t
> > happend.
>
> Are you able to get a good backtrace from the core file generated by the
> crash? Without more details, it's going to be close to impossible for
> the Asterisk developers to guess at why it's crashing for you.
>
> There's some good information at
> http://www.asterisk.org/doxygen/1.4/AstDebug.html on how to generate the
> backtrace and attach it to a bug in the bug tracker.
>
>
> --
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Digium TE120P versus Sangoma A101D-X
Paul Hales wrote: > I also understand your stand here Kevin - there is no way you can > restrict the software running on a server out in the wild, and no way to > make sure the software they are running will not conflict in any way. > > But a single port E1 card with hardware echo cancellationpossible? Hold that thought just for a little bit :-) -- Matthew Fredrickson Software/Firmware Engineer 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
[asterisk-users] Protection switching on PRIs.
Has anyone figured out a way to instantaneously swing over PRIs bearing calls in progress to another media gateway without dropping them? Obviously, this would require a DACS of some sort. But I am thinking that it is possible to swing T1s over in a DACS without actually causing the endpoint to reframe as long as the other endpoint is kept in sync. So, it'd be nice, for example, to bring in some PRIs through a small DS1-level DACS, and be able to swing them over among various boxes with quad-span Sangoma or Digium T1 cards in the event of a problem without dropping any calls. I do not imagine there is a way to do this easily from an engineering point of view; it would require constantly synchronised PRI interfaces on the hardware and intermediate (zaptel) level, and essentially massively parallel Asterisk and loads of RPC. But I imagine there is good ROI in developing something like this. Even developing a way to transparently switch SIP calls in progress (together with the media and dialog state) to another Asterisk installation seamlessly would be good, if nobody wants to go through all the other of making this work on the TDM level. Any allowance for that type of failover clustering? -- 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] Problems with Asterisk 1.4.14 and Queue app
On Thu, 2007-11-29 at 14:28 -0300, equis software wrote: > I have problems with 1.4.14, it crash every few minutes. > The same configuration and machine in Asterisk 1.4.6 it doesn´t > happend. Are you able to get a good backtrace from the core file generated by the crash? Without more details, it's going to be close to impossible for the Asterisk developers to guess at why it's crashing for you. There's some good information at http://www.asterisk.org/doxygen/1.4/AstDebug.html on how to generate the backtrace and attach it to a bug in the bug tracker. -- 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
Re: [asterisk-users] Fw: Remove a TDM Card
On Thu, Nov 29, 2007 at 11:14:12AM +0100, Sasa wrote: > Hi, my problem isn't on new voip box with latest asterisk version...my > problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this > steps for remove rightly TDM Card: > > - remove line configuration about tdm card in zapata.conf and zaptel.conf > - remove in rc.modules and rc.modules-2.4.33.3 line: > /sbin/modprobe wctdm24xxp && /sbin/ztcfg -vv > - rmmod wctdm24xxp > - halt > - remove physically card tdm from pc (box voip 1) > - restart box voip 1 You have been quite short on details. For instance: what distribution of Linux? What version of Zaptel? Do you have another Zaptel card? It seems you either have two zaphfc cards or one dual-BRI card. If so, the procedure is slightly more complicated, as you basically have to reconfigure the system afterwards. As I mentioned, genzaptelconf can be handy for that. -- Tzafrir Cohen icq#16849755 jabber:[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
[asterisk-users] SLA: Handling of errors in outgoing call
Hi All.
I've been experimenting with SLA on Asterisk 1.4.13 (patched up to
1.4.14).
I am using a SIP channel for my "trunk" line.
On the whole things are good, but I have noticed that if I misdial an
outgoing call,
i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just
drops, rather than
presenting an error tone or message to the user.
Here is my sip.conf (cleaned to protect the innocent):
register => 5000:password:[EMAIL PROTECTED]/5000
[vsp5000]
type=peer
secret=password
username=username
host=vsp.ip.address
fromuser=5000
fromdomain=vsp.domain
canreinvite=no
context=line-in
insecure=very
Here is my sla.conf:
[line1]
type=trunk
device=Local/[EMAIL PROTECTED]
[station](!) ; When there are a lot of stations that are
configured the same way,
; it is convenient to use a configuration
template like this so that
; the common settings stay in one place.
type=station
autocontext=default
trunk=line1
trunk=line2
;trunk=line3
;trunk=line4
[station1](station)
device=SIP/station1
[station2](station)
device=SIP/station2
Here is my extensions.conf:
[macro-call-sla]
; ${ARG1} - line name
exten => s,1,SLATrunk(${ARG1})
exten => s,2,Hangup
[line1_outbound]
exten => disa,1,Disa(no-password|line1_outbound)
exten => _,1,Dial(SIP/[EMAIL PROTECTED])
exten => _,2,Hangup
[line-in]
exten => 5000,1,Macro(call-sla,line1)
So to summarise:
if I seize the line and dial a number known at vsp5000 then I
get ringing etc - good.
if I seize the line and dial a number unknown at vsp5000 then
the call drops silently - not good.
Any ideas?
__
Steve Langstaff
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[asterisk-users] Problems with Asterisk 1.4.14 and Queue app
I have problems with 1.4.14, it crash every few minutes. The same configuration and machine in Asterisk 1.4.6 it doesn´t happend. Is there anybody with similiar problems? 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] least cost routing and asterisk-1.4
I should add that TransNexus's product comes with an OSP module for Asterisk that allows calls to be brokered directly through Asterisk in the dial plan logic, interfacing with the TransNexus NexSRS system. On Thu, 29 Nov 2007, Alex Balashov wrote: > > OpenSER has an LCR module. > > If you're interested in a commercial product, try www.transnexus.com. > > On Thu, 29 Nov 2007, Goke Aruna wrote: > >> Can someone guide me on what package I can use to do least cost routing >> in asterisk-1.4 without going through the prepaid calling card platforms. >> >> I have tried Asterisk::LCR and LCDial without success, if more help on >> either too. I will be glad. >> >> I will be glad for good pointers. >> >> Thanks. >> >> Goksie >> >> ___ >> --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 >> > > -- > Alex Balashov > Evariste Systems > Web: http://www.evaristesys.com/ > Tel: +1-678-954-0670 > Direct : +1-678-954-0671 > -- 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] least cost routing and asterisk-1.4
OpenSER has an LCR module. If you're interested in a commercial product, try www.transnexus.com. On Thu, 29 Nov 2007, Goke Aruna wrote: > Can someone guide me on what package I can use to do least cost routing > in asterisk-1.4 without going through the prepaid calling card platforms. > > I have tried Asterisk::LCR and LCDial without success, if more help on > either too. I will be glad. > > I will be glad for good pointers. > > Thanks. > > Goksie > > ___ > --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 > -- 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
[asterisk-users] least cost routing and asterisk-1.4
Can someone guide me on what package I can use to do least cost routing in asterisk-1.4 without going through the prepaid calling card platforms. I have tried Asterisk::LCR and LCDial without success, if more help on either too. I will be glad. I will be glad for good pointers. Thanks. Goksie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk <-> Nortel Phone Switch
It's a nortel phone switch (ie: phone company), not a nortel pbx. On Wed, Nov 28, 2007 at 08:55:09PM -0600, Jonn R Taylor wrote: > What LAN and you using? ELAN or HSP Are you trying to connect to a signaling > server? Please provide Nortel config. > > Jonn > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL > PROTECTED] > Sent: Wednesday, November 28, 2007 2:06 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk <-> Nortel Phone Switch > > Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k). > > Nortel did an upgrade which changed a bunch of things today, so I thought I'd > give it another shot. It looks like I'm much closer this time, but still no > go. Can't do calling in either direction. Anyone have any ideas? > > Thanks! > > Shawn > > > [nortel] > host=10.0.0.10 > insecure=very > type=peer > qualify=no > canreinvite=no > dtmfmode=rfc2833 > fromuser=user > username=user > secret=123 > disallow=all > allow=gsm > allow=ulaw > allow=alaw > dtmfmode=rfc2833 > usereqphone=yes > context=from-nortel > > > asterisk*CLI> sip debug ip 10.0.0.10 > SIP Debugging Enabled for IP: 10.0.0.10 > The 'sip debug' command is deprecated and will be removed in a future > release. Please use 'sip set debug' instead. > Audio is at 192.168.10.2 port 17492 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x2 (gsm) to SDP > Adding codec 0x8 (alaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to 10.0.0.10:5060: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport > From: "Shawn Ip" ;tag=as25dd7670 > To: > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Wed, 28 Nov 2007 18:24:14 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 287 > > v=0 > o=root 3386 3386 IN IP4 192.168.10.2 > s=session > c=IN IP4 192.168.10.2 > t=0 0 > m=audio 17492 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- > asterisk*CLI> > <--- SIP read from 10.0.0.10:5060 ---> > SIP/2.0 486 Busy Here > From: "Shawn Ip";tag=as25dd7670 > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > Via: SIP/2.0/UDP 192.168.10.2:5060;rport=5060;branch=z9hG4bK489024dd > User-Agent: Asterisk PBX > Max-Forwards: 70 > Supported: replaces > Date: Wed, 28 Nov 2007 18:24:14 GMT > Allow: NOTIFY > Content-Type: application/SDP > Content-Length: 287 > > v=0 > o=root 3386 3386 IN IP4 192.168.10.2 > s=session > c=IN IP4 192.168.10.2 > t=0 0 > m=audio 17492 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > <-> > --- (13 headers 14 lines) --- > Transmitting (no NAT) to 10.0.0.10:5060: > ACK sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport > From: "Shawn Ip" ;tag=as25dd7670 > o: > Contact: > Call-ID: [EMAIL PROTECTED] > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > > ___ > --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] Registration problem: UA -> SER -> Asterisk
Stefano, if you have distributed Registrars, which will keep the user location of registration ? And you do not need OpenSER to fwd Register message.. Register / Proxy / Redirect could be totally separate entities. By the way, if you post the SIP Register message likely someone could help you. "Check in Register msg Contact field" Contact: Bob ;expires=660. Probably SER replace it. Beside it, you can have a multiple OpenSER and Radius (AA) with shared db acting as Registrars, I do not understand why you want asterisk to do that. Regards, Giovanni 2007/11/29, Stefano Capitanio <[EMAIL PROTECTED]>: > > > Hi, > > Yes it make sense to have multiple registrars and to have SER acting as a > transparent proxy that forwards also REGISTER messages. > > The question is: why it does not work! L > > Regards, > Stefano > > > > > > Stefano, > It is not Asterisk, It is SER (dispatcher module ?). > Why Asterisk is acting as Register ? make sense use openSER as > Register/Proxy and Asterisk only Proxy and MG > > Regards, > Giovanni > > > > ___ > --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 > -- Giovanni Miano ___ --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] roundrobin and rrmemory with pre-defined agent order
I've also looked into this issue, and it seems that asterisk doesn't respect the order of the members in queues.conf. Asterisk uses a hash table internally to hold the queue members. I guess it's fine when you have dozens of agents, but for simpler scenarios, it's a pain not to be able to determine agent's order. Julian J. M. On Nov 29, 2007 1:46 PM, Fernando Urzedo <[EMAIL PROTECTED]> wrote: > > Hi, > > I would like to implement a queue using a circular strategy, I mean, > using roundrobin or rrmemory strategies. However, I am not able to > define the order Asterisk will call the agents once a new call arrives > in the queue. Seems that Asterisk will always define its order as the > queues.conf file is read, and most of times this order is different from > the one I want (for each queue in queues.conf, I add members in the > order I want them to be called). > > I tried to use the "penalty" setting, but then Asterisk gets stuck in > the first agent (lowest penalty) until it answers a call. > > Is there a way to implement what I am trying? I am using Asterisk > 1.2.19... > > Thanks in advance! -- http://www.julianmenendez.es ___ --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] Sangoma Question
Jeremy Mann wrote: > And they work with Asterisk/Zaptel 1.4 ? > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson > Sent: Wednesday, November 28, 2007 11:06 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Sangoma Question > > On Nov 28, 2007 10:52 AM, Jeremy Mann <[EMAIL PROTECTED]> wrote: > >> Do sangoma cards use the standard Zaptel drivers? Or do they have to be >> compiled externally like Rhino cards? >> > > Sangoma maintains a patchset that gets applied to the stock zaptel > drivers before compilation. They provide automated tools that will > take care of the patching/compiling/installing/configuring for you. > > -erik > The only issue I have seen with the new improved easy installation script is if you are doing something a little different such as AMI,D4. You will need to run wancfg and change the automagic settings manually. Thanks, Steve Totaro ___ --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] G729/MOH Quality
Steve Underwood wrote: > Darryl Dunkin wrote: >> Does anyone have any opinions on the music on hold quality over G729? >> The stock files seem to sound terrible over it, this is enhanced further >> by calls coming from the PSTN via a Zaptel gateway. I am only using the >> stock wav files and have not attempted to use much else so far. >> >> I've ruled out timing issues on the system generating the MOH itself >> (ztdummy on the PBX itself, our Zaptel gateway is a separate Asterisk >> server). There is no transcoding going on in the middle except via our >> Zaptel/T1 gateway. When using G711 it sounds fine of course, but this >> doesn't work well for remote sites with lower bandwidth connections. >> >> As of now, I'm torn between getting complaints from end users about the >> music or killing it entirely (leaving people waiting in queues with dead >> silence). >> > > Music over G.729, or any other similarly low bit rate codec sounds > bloody awful. The only thing you can do to mitigate this is choose > simpler music. A single voice singing sounds OK - that's what the codec > is optimsed for. A singer with a guitar doesn't usually sounds too bad. > As the music gets more complex it goes horribly downhill from there. > > Steve I've seen the same problem. Currently i have no need for that, but maybe anybody knows - where to get something more-or-less classic (like bundled MoH) for free or small price. Regards, Atis ___ --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] roundrobin and rrmemory with pre-defined agent order
Hi, I would like to implement a queue using a circular strategy, I mean, using roundrobin or rrmemory strategies. However, I am not able to define the order Asterisk will call the agents once a new call arrives in the queue. Seems that Asterisk will always define its order as the queues.conf file is read, and most of times this order is different from the one I want (for each queue in queues.conf, I add members in the order I want them to be called). I tried to use the "penalty" setting, but then Asterisk gets stuck in the first agent (lowest penalty) until it answers a call. Is there a way to implement what I am trying? I am using Asterisk 1.2.19... Thanks in advance! LocaWeb Telecom www.locawebtelecom.com.br :: Central Suporte 24 horas :: http://www.locawebtelecom.com.br/suporte ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 40, Issue 82
i am using TDM400P (clone) > Message: 22 > Date: Thu, 29 Nov 2007 10:20:12 +0800 > From: "zhao bo" <[EMAIL PROTECTED]> > Subject: [asterisk-users] FSK signal start after second ring > To: asterisk-users@lists.digium.com > Message-ID: ><[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > hi all > my local PSTN use an different cid signalling format > [ring-DTMF-ring],and my asterisk server can not get the cid > info,allways say "UNKNOWN".I try to use EX220 (caller id converter) > translate DTMF into FSK,but after that the signal looks like > [ring-DTMF-ring-FSK],asterisk still "UNKNOWN" .Is there someway that i > can tell asterisk that my fsk signalling start after second ring ? > > sorry for my English! ___ --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 TE120P versus Sangoma A101D-X
Paul Hales wrote: > But a single port E1 card with hardware echo cancellationpossible? Yes, I would say that is definitely possible (wink wink). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ___ --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] What is voice format 8
Moises Silva wrote: > You should not care for debug messages unless you are debugging. > I have been debugging. My IAXmodem connection. > "core show codecs" > > Will show you format 8 is ALAW > thanks. > - Moy > > On Nov 28, 2007 2:41 PM, Robert Moskowitz <[EMAIL PROTECTED]> wrote: > >> The IAX2 channel is to IAXmodem. >> The SIP extension is an ATA with a fax attached. >> >> >> Nov 28 15:30:20 DEBUG[2997] chan_sip.c: build_route: Contact hop: >> Nov 28 15:30:20 VERBOSE[3276] logger.c: -- SIP/2201-090995f0 answered >> IAX2/24729-2 >> Nov 28 15:30:20 DEBUG[2995] chan_iax2.c: Ooh, voice format changed to 8 >> >> So what does this mean? >> >> The fax works just fine. I am just trying to tune up my dialplan. >> >> >> >> ___ >> --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] To DB or not to DB?
Thanks to all the people who commented. It sounds like I don't really need it currently, but worth experimenting with for larger deployments. Cheers Alan -- The way out is open! http://www.theopensourcerer.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
[asterisk-users] Registration problem: UA -> SER -> Asterisk
Hi, Yes it make sense to have multiple registrars and to have SER acting as a transparent proxy that forwards also REGISTER messages. The question is: why it does not work! :-( Regards, Stefano Stefano, It is not Asterisk, It is SER (dispatcher module ?). Why Asterisk is acting as Register ? make sense use openSER as Register/Proxy and Asterisk only Proxy and MG Regards, Giovanni ___ --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] Shared line appearance phones?
Russell Bryant wrote: > Ron McCarthy wrote: >> Asterisk 1.4 im guessing? I did not know the Snom's worked with that, >> Ill have to check it out then! > > The way it is implemented in Asterisk is a bit interesting. It uses the > existing device state support (hints, BLF) to manage the buttons for shared > lines. Asterisk changes the state of these virtual "shared lines" to > different > states, and the light on the phone reflects the state (in use, ringing, on > hold). I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID from a PRI on several Polycom IP430's. Might you be willing to share some specific configurations for such a situation? thanks mark ___ --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] Hylafax
Hi, We seem to be having some teething issues with a new Hylafax - happy to pay someone to complete the installation. Please contact offlist. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499 ___ --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] G729/MOH Quality
Darryl Dunkin wrote: > Does anyone have any opinions on the music on hold quality over G729? > The stock files seem to sound terrible over it, this is enhanced further > by calls coming from the PSTN via a Zaptel gateway. I am only using the > stock wav files and have not attempted to use much else so far. > > I've ruled out timing issues on the system generating the MOH itself > (ztdummy on the PBX itself, our Zaptel gateway is a separate Asterisk > server). There is no transcoding going on in the middle except via our > Zaptel/T1 gateway. When using G711 it sounds fine of course, but this > doesn't work well for remote sites with lower bandwidth connections. > > As of now, I'm torn between getting complaints from end users about the > music or killing it entirely (leaving people waiting in queues with dead > silence). > Music over G.729, or any other similarly low bit rate codec sounds bloody awful. The only thing you can do to mitigate this is choose simpler music. A single voice singing sounds OK - that's what the codec is optimsed for. A singer with a guitar doesn't usually sounds too bad. As the music gets more complex it goes horribly downhill from there. 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] Fw: Remove a TDM Card
Hi, This procedure should work whether you have remove some "VoIP" card from your VoIP box. Anyway be careful On Nov 29, 2007 11:14 AM, Sasa <[EMAIL PROTECTED]> wrote: > Hi, my problem isn't on new voip box with latest asterisk version...my > problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this > steps for remove rightly TDM Card: > > - remove line configuration about tdm card in zapata.conf and zaptel.conf > - remove in rc.modules and rc.modules-2.4.33.3 line: > /sbin/modprobe wctdm24xxp && /sbin/ztcfg -vv > - rmmod wctdm24xxp > - halt > - remove physically card tdm from pc (box voip 1) > - restart box voip 1 > > ..this procedure is ok ? > Thanks ! > > -- > > Salvatore. > > > > - Original Message - > From: "Tzafrir Cohen" <[EMAIL PROTECTED]> > To: > Sent: Thursday, November 29, 2007 1:50 AM > Subject: Re: [asterisk-users] Fw: Remove a TDM Card > > > > On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote: > >> Hi, sorry but perhaps I don't have explained clearly my problem...now I > >> have > >> a box voip that must be replace with another box voip but I want to do > >> test > >> before remove the old voip from production. > > > > With later versions of Zaptel you have zapconf and genzaptelconf . Use > > either of them to generate /etc/zaptel.conf and to generate a sample > > zapata.conf snippet in /etc/asterisk/zapata-channels.conf . > > > > -- > > Tzafrir Cohen > > icq#16849755 jabber:[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 > > > > > ___ > --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] Fw: Remove a TDM Card
Hi, my problem isn't on new voip box with latest asterisk version...my problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this steps for remove rightly TDM Card: - remove line configuration about tdm card in zapata.conf and zaptel.conf - remove in rc.modules and rc.modules-2.4.33.3 line: /sbin/modprobe wctdm24xxp && /sbin/ztcfg -vv - rmmod wctdm24xxp - halt - remove physically card tdm from pc (box voip 1) - restart box voip 1 ..this procedure is ok ? Thanks ! -- Salvatore. - Original Message - From: "Tzafrir Cohen" <[EMAIL PROTECTED]> To: Sent: Thursday, November 29, 2007 1:50 AM Subject: Re: [asterisk-users] Fw: Remove a TDM Card > On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote: >> Hi, sorry but perhaps I don't have explained clearly my problem...now I >> have >> a box voip that must be replace with another box voip but I want to do >> test >> before remove the old voip from production. > > With later versions of Zaptel you have zapconf and genzaptelconf . Use > either of them to generate /etc/zaptel.conf and to generate a sample > zapata.conf snippet in /etc/asterisk/zapata-channels.conf . > > -- > Tzafrir Cohen > icq#16849755 jabber:[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 > ___ --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] Needed Hardware
Hi All; I would like the needed hardware (MHz, MB, and GBI) for the following: 1) Users: 30 IP Phones. 2) IP Trunk for maximum 10 concurrent calls, with g729 codec. 3) Analogue card of 8 lines FXO. 4) Softphone 5 and they use g729 codec. 5) Functions to be normal functions (call pickup, call forward, call trasnfsre, bridges, whisper) and for 60% of the users. Any advise? Regards Bilal Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now. http://mobile.yahoo.com/sports;_ylt=At9_qDKvtAbMuh1G1SQtBI7ntAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk API Manager
It could be a good idea. Right now, I'm trying to verify it's possible to identify a caller and his postion in queue with his URI after having got the result from the command sent to Asterisk Manager. Scott Wolfe a écrit : > Write a application to log the information to a DB, then have all other > clients connect to the database for the status. Unless I am missing > something. > > -Scott > > - Original Message - > From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, November 28, 2007 11:29 AM > Subject: Re: [asterisk-users] Asterisk API Manager > > > If each client connected only once, subsequently made a request every > minute, and then disconnected only when finished, the load might be more > reasonable. It can be a little harder to write that kind of client > though :) > > Mojo > > Mojo with Horan & Company, LLC wrote: > >> So you'd be making 100 connections/minute, which is pretty relentless. >> That's like five connections, five requests sent, five responses >> received, and five disconnects, /every/ three seconds. >> >> And the likelihood of all 100 users to be spread out evenly over a >> minute doesn't seem very high. I think your box would be pretty busy >> with that. >> >> Astmanproxy would be indicated. >> >> Moj >> >> >> Anthony Chapellier wrote: >> >> >>> However I wanted to get periodic infos about queued users (position in >>> queue) only. So I thought I could make a program sending periodic >>> requests to asterisk manager. Is it really bad to bother asterisk >>> manager with frequently and periodic requests sent by mutiple users (we >>> could say maybe 100 users with a request every minute) ? >>> >>> Moises Silva a écrit : >>> >>> >>> Yes, but you should use astmanproxy instead and don't bother Asterisk with multiple manager connections. On Nov 27, 2007 8:24 AM, Anthony Chapellier <[EMAIL PROTECTED]> wrote: > Hi, > > Does Asterisk manager allow multiple clients to connect to an Asterisk > instance using the same user account ? > > 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 > > > > > >>> ___ >>> --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 > > > ___ > --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] DTMF not recognized on ISDN with Siemens -not IP- phone
Paul Hales a écrit : > The current Digium BRI cards need the phones to send DTMF over as > SIP-INFO. > Not sure why, but googling should help. (I think this is even covered on > the Digium site) > Hi Paul, don't quite understand what you mean: how can a regular phone (not IP) send SIP Info DTMF? Do you mean we should add a parameter in /etc/misdn-init.conf or /etc/asterisk/misdn.conf? Meanwhile, we only have the problem with Siemens phone, others brand are ok. Thanks for your help. > On Wed, 2007-11-28 at 09:47 +0100, Administrator TOOTAI wrote: > >> Good day all, >> >> we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244 >> with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the >> office through ISDN lines and then get a possibility to join meetme >> conference. It works well except when customers are using SIEMENS phones >> (not IP): DTMF is not recognized. >> >> Does someone have an idea on what could be the problem with those phones? >> >> Thanks for any help >> >> -- Daniel ___ --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] CDR n Dial A option
When I use Dial(type/identifier, timeout, A(some_file)) CDR billsec starts when announcement ends. But I have to bill from when called party answers to phone. How can I solve my problem? -- Suich ___ --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
