RE: [Asterisk-Users] voicemail file access problems
> > On Wed, 2003-07-30 at 22:33, Patrick wrote: > > > Did it work after you left a new voice mail message? > > > > > > I was looking into the source code to fix it so that the euid was set to > > > nobody, create the file and then change it back to uid 0, but that didn't > > > work. Or, maybe change the file mode was 770 with the group set so that > > > the webserver could modify the file so I wouldn't have to run a suid .cgi > > > script. > > > > If you create the _directories_ the files are going to be created in > > with group apache (or whatever group your webserver runs under), with > > the sgid bit set, doesn't that cause the file to be created with proper > > permission for the cgi? > But the mask of the file is set to 0700. I don't think the sgid bit will > make a difference if the file isn't written 0770. It's still on > readable/writable/executable by the owner. ACK. I read over that. But I think that combining the sgid creation of directories and changing 0700 to 2770 for the mkdir calls, and 0700 to 0700 for the call to ast_writefile should get you where you want to be. I haven't tested it, it just _looks_ that way. (assumption is the mother of all fuckups) good luck, Armand. -- A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing on usenet and in email? signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Call Transfer
Excellent idea mate, Now I am able to do what I wanted with Great help from Jeremy McNamara. Thanks alot Foong - Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer > Foong > > Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below > > Channel: SIP/[EMAIL PROTECTED] > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > Context: mysipcontext2 > Extension: 2000 > Priority: 1 > > This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. > > All you need is a script to lookup in the database and generate the script file for you and it's done. > > HTH > > Andy > > > *** REPLY SEPARATOR *** > > On 30/07/2003 at 16:30 Chee Foong wrote: > > >Hello Dan, > > > >Thanks for you reply. > > > >Base on you recomendation using the 'T' argument. I manage to do call > >transfer an it works really well. > > > >My problem comes when my boss comes out with a superb idea where the > >transfering process is automated without involving a human :( > > > >Say asterisk get 2 numbers (from database, text file, etc), one belongs > >party A and the other belongs to party B. Asterisk will calls both parties > >and do the tranfer automatically. In another words, asterisk is resposible > >to 'press' the '#' to do the transfer. I don't this can be achieve in the > >extension.conf not matter how you structure you dial plan. > > > >Perhaps, the only way is to write a apps and plug it into asterisk like all > >the asterisk modules such as Meetme. > > > >Any ideas? > > > > > >Foong > > > >- Original Message - > >From: "Dan" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Wednesday, July 30, 2003 3:42 PM > >Subject: Re: [Asterisk-Users] Call Transfer > > > > > >> Hi, > >> > >> It works if you put the 'T' switch in the dial line. > >> > >> You can then transfer the call from the caller. > >> I have tested it in the folllowing configuration and it works: > >> Call from a Cisco 7960 to an ATA 186. > >> Select 'Transfer" on 7960 > >> Call another extension (X-Lite) > >> Select again transfer on 7960. > >> The call remain between ATA and X-Lite. > >> > >> This is what you need? > >> > >> BR, > >> Dan > >> > >> - Original Message - > >> From: "Chee Foong" <[EMAIL PROTECTED]> > >> To: <[EMAIL PROTECTED]> > >> Sent: Wednesday, July 30, 2003 7:08 AM > >> Subject: [Asterisk-Users] Call Transfer > >> > >> > >> Hello all, > >> > >> I am in a situation where I need to use asterisk to call someone say > >Party > >> A. After the call to Party A got through, asterisk will put Party A on > >hold, > >> then asterisk will call Party B. If call to Party B got through, asterisk > >> will transfer Party A to Party B. > >> > >> I wonder if this features is implemented into asterisk. I have found a > >post > >> in asterisk mailing list: > >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html > >> > >> but that doesn't help much. > >> > >> If this features is not implemented, can anyone give me some point on how > >to > >> implement this in asterisk? Do I need to write an app like the Dial apps > >for > >> asterisk to load at start up? > >> > >> > >> thanks > >> > >> Foong > >> > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Inbound calls work pretty fine again,
Thanks for you help
Yves
|-+->
| | "Brenton D. Rothchild"|
| | <[EMAIL PROTECTED]> |
| | Sent by: |
| | [EMAIL PROTECTED]|
| | .digium.com |
| | |
| | |
| | 30/07/2003 16:15 |
| | Please respond to |
| | asterisk-users|
| | |
|-+->
>---|
|
|
| To: <[EMAIL PROTECTED]>
|
| cc:
|
| Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
|
>---|
That also worked for me. My AudioCodes MP-104 FXO has no problem
making inbound calls now.
Thanks Patrick and Adam.
-Brenton
- Original Message -
From: "Low, Adam" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 8:45 AM
Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
> Well found Patrick, that did the trick for me as well !
>
> I had been trying to debug 1.135 where this portion of code wasn't added
yet ... thats a lesson learnt ...
>
> -Original Message-
> From: Patrick
> To: '[EMAIL PROTECTED] '
> Sent: 30/07/03 15:04
> Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
>
>
> It is in the find_user() routine. If it is not an extension on the
> PBX,
> it should return a zero
>
> if ( isfound ) {
>ast_log(LOG_DEBUG, "%s is not a local user\n", name);
>ast_pthread_mutex_unlock(&userl.lock);
>return 1; <--- this is the problem - change it to a 0.
> }
>
> It isn't an error, so it should just return. Change that and the
> function
> will work properly. I tested it using an AS5350 and successly made an
> inbound call.
>
> Patrick
>
>
> On Wed, 30 Jul 2003, Low, Adam wrote:
>
> > Brenton, Yves, ...
> >
> > I've located the cause of the problem in chan_sip.c but am still
> trying to find the exact cause being completely new to the asterisk
> code. It seems that there was an added function in 1.135 called
> 'find_user' that is supposed to lookup the users incoming call limit but
> the routine is unable to find a matching user for my AS5300 which I
> suspect is because it does not REGISTER with the server prior to
> attempting to send calls.
> >
> > I'm going to continue debugging a little later and see if I can narrow
> it down more ...
> >
> > Adam
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > To: [EMAIL PROTECTED]
> > Sent: 30/07/03 14:09
> > Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs
> 1.134
> >
> >
> > Hi,
> >
> > I am using the latest cvs release of asterisk, and the behaviour is in
> > fact
> > the same,
> >
> > outbound calls work fine,
> > but for inbound calls (from C2651 over PSTN) , SIP messages get
> > "blocked"
> > by asterisk, and never reach the phone.
> >
> > The setup is the same : 7960 <--> asterisk <--> C2651<->
> > PSTN
> >
> > Yves
> >
> >
> > |-+->
> > | | "Low, Adam" |
> > | | <[EMAIL PROTECTED]>|
> > | | Sent by: |
> > | | [EMAIL PROTECTED]|
> > | | .digium.com |
> > | | |
> > | | |
> > | | 30/07/2003 11:37 |
> > | | Please respond to |
> > | | asterisk-users|
> > | | |
> > |-+->
> >
> >
> >---
> > |
> > |
> > |
> > | To: "'[EMAIL PROTECTED]'"
> > <[EMAIL PROTECTED]> |
> > | cc:
> > |
> > | Subject: [Asterisk-Users] chan_sip.c problems problems from
> > cvs 1.134 |
> >
> >
> >---
> > -
Re: [Asterisk-Users] Manager.pm port
On Wed, 2003-07-30 at 21:59, Steven J. Sobol wrote: > For anyone that cares... > > I am porting James Golovich's Manager.pm over to PHP. I plan on also > doing some documentation which will cover both the Perl and PHP APIs, > which will be almost identical (at least, to whatever extent is > practical). If you are running the manager from the webpage, then I can remotely understand php manager interface. But if you plan on making a command line manager app, then please do yourself a favor and just help with the perl stuff. Remember php is perl -1 or more revisions. I do say this as a php programmer for my work. We don't let php infect our backend console apps. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCO/Linux concerns
On Wed, 2003-07-30 at 18:07, Ajit M Kallingal wrote: > Hello > Since I am getting a bit concerned about the SCO vs IBM issue, I was > wondering if can I can setup Asterisk on FreeBSD is it supported ? > Are drivers for Digium cards available on FreeBSD ? If you are worried about it, you really should look into what happened during the lawsuit between Bellcore/USL and UC berkely. Basically they removed all the offending parts from BSD and then rewrote them from scratch. That is why there is a 4.4 lite BSD, it was they after settlement release. It wasn't a full unix anymore, but it had enough to start from again. From that core, you get the *BSD systems. The worst that will happen is a judge will deem certain parts as infringing and order them removed. At which time the functionality will be replaced by originally written code. No problems. So far the only chance of finding infringing code from SCO to linux was donated by a caldera employee. And do remember Caldera purchased the Unix code and then changed names to the SCO group. It isn't even the people who used to write and maintain it. The real fun is when the judge goes in and finds the infringing code in SCO's linux compatibility code that caldera employees have rumored about. If the GPL is enforceable, then all of the historical unix code would fall under GPL then and the lawsuit will blow away in a poof of logic. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
With shipping, I recall my 102 came to $97. I think it was $85 but I'd need to look it up and don't have the papers nearby. -reed At 06:39 PM 7/30/2003 -0500, you wrote: I was quoted $75 and $85 USD today. Ricardo Villa http://www.telesip.net - Original Message - From: "Joe Cooke" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:31 PM Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 & 102 > I was quoted the $75 and $85 USD prices from Grandstream direct about 2 > months ago. I'm not sure if it makes a difference, but I live in the US. > > - Joe > - Original Message - > From: "marrandy" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 7:17 PM > Subject: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > > > > > Checking the earlier mails, it stated that the phones were $75 (100) & $85 > > (102) ref :- > > > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html > > > > Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the > person > > said there was no price change. > > > > Anyone on this list actually bought them at the $75 & $85 rate ??? > > > > Regards...Martin > > -- > > Too much is just enough. > > -- Mark Twain, on whiskey > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip -> h323 -> ptsn
using chan_h323 and g.711u bkw On Wed, 30 Jul 2003, Patrick wrote: > > Which codec are you using? and which H.323 channel driver? chan_h323 or > chan_oh323 ? > > > On 30 Jul 2003, Eric Wieling wrote: > > > That only works if you are using the G711 (ulaw/alaw) codecs. Other > > codecs distort inband DTMF. > > > > On Wed, 2003-07-30 at 15:26, Patrick wrote: > > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > > > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > > > I have this setup: > > > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > > > Sip phones are setup for out of band dtmf > > > > > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > > > the sip phones to the ptsn via the h323 gateway? > > > > > > > > bkw > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
http://store.yahoo.com/grandstream-networks-inc/products.html I think that will clear it up. On Wed, 30 Jul 2003, Ricardo Villa wrote: > I was quoted $75 and $85 USD today. > > Ricardo Villa > http://www.telesip.net > > - Original Message - > From: "Joe Cooke" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 6:31 PM > Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > > > I was quoted the $75 and $85 USD prices from Grandstream direct about 2 > > months ago. I'm not sure if it makes a difference, but I live in the US. > > > > - Joe > > - Original Message - > > From: "marrandy" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Wednesday, July 30, 2003 7:17 PM > > Subject: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > > > > > > > > > Checking the earlier mails, it stated that the phones were $75 (100) & > $85 > > > (102) ref :- > > > > > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html > > > > > > Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the > > person > > > said there was no price change. > > > > > > Anyone on this list actually bought them at the $75 & $85 rate ??? > > > > > > Regards...Martin > > > -- > > > Too much is just enough. > > > -- Mark Twain, on whiskey > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco5300 with asterisk through H323
Dear all, Can some of you give us some suggestion how to configure the asterisk in order to make a call to cisco5300 in g729a codec. And how to confiure the cisco5300 part in order to receive a call from cisco5300 via h323 g729a. Your advice /help will be highly appreciated. Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault
Hi Jeremy, Ok, still learning to get the backtrace. will post a trace next. When I issues a dial command on console, ex dial H323/6031334000 I get seg fault also, this only happen if it involve dialing through H323 channels Thank for your reply Foong - Original Message - From: "Jeremy McNamara" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 31, 2003 11:07 AM Subject: Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault > Send me the backtrace and console output, off list. > > That's a pretty crazy extension. I bet your trying to make some kind > of crazy callback system :) > > > > Jeremy McNamara > > > > > Chee Foong wrote: > > >I dumped the following test.call file into /var/spool/asterisk/outgoing > >gives me segmentation fault :( > > > >Channel: H323/0143126544 > >MaxRetries: 2 > >RetryTime: 60 > >WaitTime: 30 > >Context: voip-test > >Extension: 90324324433 > >Priority: 1 > > > >same thing happend if I execute dial command on console. > > > >I figure out that this happen only if I dial through a H323 channel. I am > >using chan_h323. > > > >Any one experience the same thing? > > > >Foong > > > >- Original Message - > >From: "Andy Powell" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Wednesday, July 30, 2003 6:56 PM > >Subject: Re: [Asterisk-Users] Call Transfer > > > > > > > > > >>Foong > >> > >>Take a look at the sample.call file, modifying the settings in there and > >> > >> > >copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial > >the call.. an example config is below > > > > > >>Channel: SIP/[EMAIL PROTECTED] > >>MaxRetries: 2 > >>RetryTime: 60 > >>WaitTime: 30 > >>Context: mysipcontext2 > >>Extension: 2000 > >>Priority: 1 > >> > >>This will make asterisk dial exten 1000 in the context mysipcontext when > >> > >> > >it's answered it will then call exten 2000 in mysipcontext2.. > > > > > >>All you need is a script to lookup in the database and generate the script > >> > >> > >file for you and it's done. > > > > > >>HTH > >> > >>Andy > >> > >> > >>*** REPLY SEPARATOR *** > >> > >>On 30/07/2003 at 16:30 Chee Foong wrote: > >> > >> > >> > >>>Hello Dan, > >>> > >>>Thanks for you reply. > >>> > >>>Base on you recomendation using the 'T' argument. I manage to do call > >>>transfer an it works really well. > >>> > >>>My problem comes when my boss comes out with a superb idea where the > >>>transfering process is automated without involving a human :( > >>> > >>>Say asterisk get 2 numbers (from database, text file, etc), one belongs > >>>party A and the other belongs to party B. Asterisk will calls both > >>> > >>> > >parties > > > > > >>>and do the tranfer automatically. In another words, asterisk is > >>> > >>> > >resposible > > > > > >>>to 'press' the '#' to do the transfer. I don't this can be achieve in the > >>>extension.conf not matter how you structure you dial plan. > >>> > >>>Perhaps, the only way is to write a apps and plug it into asterisk like > >>> > >>> > >all > > > > > >>>the asterisk modules such as Meetme. > >>> > >>>Any ideas? > >>> > >>> > >>>Foong > >>> > >>>- Original Message - > >>>From: "Dan" <[EMAIL PROTECTED]> > >>>To: <[EMAIL PROTECTED]> > >>>Sent: Wednesday, July 30, 2003 3:42 PM > >>>Subject: Re: [Asterisk-Users] Call Transfer > >>> > >>> > >>> > >>> > Hi, > > It works if you put the 'T' switch in the dial line. > > You can then transfer the call from the caller. > I have tested it in the folllowing configuration and it works: > Call from a Cisco 7960 to an ATA 186. > Select 'Transfer" on 7960 > Call another extension (X-Lite) > Select again transfer on 7960. > The call remain between ATA and X-Lite. > > This is what you need? > > BR, > Dan > > - Original Message - > From: "Chee Foong" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 7:08 AM > Subject: [Asterisk-Users] Call Transfer > > > Hello all, > > I am in a situation where I need to use asterisk to call someone say > > > >>>Party > >>> > >>> > A. After the call to Party A got through, asterisk will put Party A on > > > >>>hold, > >>> > >>> > then asterisk will call Party B. If call to Party B got through, > > > >asterisk > > > > > will transfer Party A to Party B. > > I wonder if this features is implemented into asterisk. I have found a > > > >>>post > >>> > >>> > in asterisk mailing list: > http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html > > but that doesn't help much. > > If this features is not implemented, can anyone give me some point on > > > >how > > > > > >>>to > >>> > >>> > implement this in asterisk? Do I need to write an app like the Dial > > > >apps > > > > > >>>for > >>> > >>> > asterisk to load at start up? > >>
RE: [Asterisk-Users] SCO/Linux concerns
What's your concern with it? If any of SCO code made it into GNU stuff, it will be removed and rewritten in a short time anyway... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ajit M Kallingal Sent: Wednesday, July 30, 2003 7:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SCO/Linux concerns Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users digest, Vol 1 #935 - 14 msgs > Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > >1. RE: voicemail file access problems (Todd Lieberman) >2. sip -> h323 -> ptsn (Brian West) >3. RE: voicemail file access problems (Todd Lieberman) >4. Re: voicemail file access problems (Tilghman Lesher) >5. Re: sip -> h323 -> ptsn (Patrick) >6. RE: voicemail file access problems (Patrick) >7. Re: sip -> h323 -> ptsn (Brian West) >8. Re: sip -> h323 -> ptsn (Patrick) >9. X100P and incoming Context + CDR? (Darren Smith) > 10. Re: CVS Problem? (Kyle Hagan) > 11. Re: sip -> h323 -> ptsn (Eric Wieling) > 12. %unsuscribe (Carlos Crembil) > 13. Re: SetCIDName (Siggi Langauf) > 14. RE: X-Lite and Call transfer using Asterisk (Stuart Hirst) > > --__--__-- > > Message: 1 > From: "Todd Lieberman" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:49:56 -0400 > Reply-To: [EMAIL PROTECTED] > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 2 > Date: Wed, 30 Jul 2003 15:08:53 -0500 (CDT) > From: Brian West <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: [EMAIL PROTECTED] > > I have this setup: > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > Sip phones are setup for out of band dtmf > > but the h323 gateway is inband. Is their a way to pass the digits from > the sip phones to the ptsn via the h323 gateway? > > bkw > > --__--__-- > > Message: 3 > From: "Todd Lieberman" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 16:12:59 -0400 > Reply-To: [EMAIL PROTECTED] > > I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. > > you still need to make sure nobody has read/write permission on > /var/spool/asterisk/vm/$MBOX > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Ma
Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault
Send me the backtrace and console output, off list. That's a pretty crazy extension. I bet your trying to make some kind of crazy callback system :) Jeremy McNamara Chee Foong wrote: I dumped the following test.call file into /var/spool/asterisk/outgoing gives me segmentation fault :( Channel: H323/0143126544 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: voip-test Extension: 90324324433 Priority: 1 same thing happend if I execute dial command on console. I figure out that this happen only if I dial through a H323 channel. I am using chan_h323. Any one experience the same thing? Foong - Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: Hello Dan, Thanks for you reply. Base on you recomendation using the 'T' argument. I manage to do call transfer an it works really well. My problem comes when my boss comes out with a superb idea where the transfering process is automated without involving a human :( Say asterisk get 2 numbers (from database, text file, etc), one belongs party A and the other belongs to party B. Asterisk will calls both parties and do the tranfer automatically. In another words, asterisk is resposible to 'press' the '#' to do the transfer. I don't this can be achieve in the extension.conf not matter how you structure you dial plan. Perhaps, the only way is to write a apps and plug it into asterisk like all the asterisk modules such as Meetme. Any ideas? Foong - Original Message - From: "Dan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 3:42 PM Subject: Re: [Asterisk-Users] Call Transfer Hi, It works if you put the 'T' switch in the dial line. You can then transfer the call from the caller. I have tested it in the folllowing configuration and it works: Call from a Cisco 7960 to an ATA 186. Select 'Transfer" on 7960 Call another extension (X-Lite) Select again transfer on 7960. The call remain between ATA and X-Lite. This is what you need? BR, Dan - Original Message - From: "Chee Foong" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 7:08 AM Subject: [Asterisk-Users] Call Transfer Hello all, I am in a situation where I need to use asterisk to call someone say Party A. After the call to Party A got through, asterisk will put Party A on hold, then asterisk will call Party B. If call to Party B got through, asterisk will transfer Party A to Party B. I wonder if this features is implemented into asterisk. I have found a post in asterisk mailing list: http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html but that doesn't help much. If this features is not implemented, can anyone give me some point on how to implement this in asterisk? Do I need to write an app like the Dial apps for asterisk to load at start up? thanks Foong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager.pm port
For anyone that cares... I am porting James Golovich's Manager.pm over to PHP. I plan on also doing some documentation which will cover both the Perl and PHP APIs, which will be almost identical (at least, to whatever extent is practical). Will let y'all know when I have some usable code to show you. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sample.call + chan_h323 gives seg fault
I dumped the following test.call file into /var/spool/asterisk/outgoing gives me segmentation fault :( Channel: H323/0143126544 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: voip-test Extension: 90324324433 Priority: 1 same thing happend if I execute dial command on console. I figure out that this happen only if I dial through a H323 channel. I am using chan_h323. Any one experience the same thing? Foong - Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer > Foong > > Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below > > Channel: SIP/[EMAIL PROTECTED] > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > Context: mysipcontext2 > Extension: 2000 > Priority: 1 > > This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. > > All you need is a script to lookup in the database and generate the script file for you and it's done. > > HTH > > Andy > > > *** REPLY SEPARATOR *** > > On 30/07/2003 at 16:30 Chee Foong wrote: > > >Hello Dan, > > > >Thanks for you reply. > > > >Base on you recomendation using the 'T' argument. I manage to do call > >transfer an it works really well. > > > >My problem comes when my boss comes out with a superb idea where the > >transfering process is automated without involving a human :( > > > >Say asterisk get 2 numbers (from database, text file, etc), one belongs > >party A and the other belongs to party B. Asterisk will calls both parties > >and do the tranfer automatically. In another words, asterisk is resposible > >to 'press' the '#' to do the transfer. I don't this can be achieve in the > >extension.conf not matter how you structure you dial plan. > > > >Perhaps, the only way is to write a apps and plug it into asterisk like all > >the asterisk modules such as Meetme. > > > >Any ideas? > > > > > >Foong > > > >- Original Message - > >From: "Dan" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Wednesday, July 30, 2003 3:42 PM > >Subject: Re: [Asterisk-Users] Call Transfer > > > > > >> Hi, > >> > >> It works if you put the 'T' switch in the dial line. > >> > >> You can then transfer the call from the caller. > >> I have tested it in the folllowing configuration and it works: > >> Call from a Cisco 7960 to an ATA 186. > >> Select 'Transfer" on 7960 > >> Call another extension (X-Lite) > >> Select again transfer on 7960. > >> The call remain between ATA and X-Lite. > >> > >> This is what you need? > >> > >> BR, > >> Dan > >> > >> - Original Message - > >> From: "Chee Foong" <[EMAIL PROTECTED]> > >> To: <[EMAIL PROTECTED]> > >> Sent: Wednesday, July 30, 2003 7:08 AM > >> Subject: [Asterisk-Users] Call Transfer > >> > >> > >> Hello all, > >> > >> I am in a situation where I need to use asterisk to call someone say > >Party > >> A. After the call to Party A got through, asterisk will put Party A on > >hold, > >> then asterisk will call Party B. If call to Party B got through, asterisk > >> will transfer Party A to Party B. > >> > >> I wonder if this features is implemented into asterisk. I have found a > >post > >> in asterisk mailing list: > >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html > >> > >> but that doesn't help much. > >> > >> If this features is not implemented, can anyone give me some point on how > >to > >> implement this in asterisk? Do I need to write an app like the Dial apps > >for > >> asterisk to load at start up? > >> > >> > >> thanks > >> > >> Foong > >> > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
But the mask of the file is set to 0700. I don't think the sgid bit will make a difference if the file isn't written 0770. It's still on readable/writable/executable by the owner. Patrick On 31 Jul 2003, Armand A. Verstappen wrote: > On Wed, 2003-07-30 at 22:33, Patrick wrote: > > Did it work after you left a new voice mail message? > > > > I was looking into the source code to fix it so that the euid was set to > > nobody, create the file and then change it back to uid 0, but that didn't > > work. Or, maybe change the file mode was 770 with the group set so that > > the webserver could modify the file so I wouldn't have to run a suid .cgi > > script. > > If you create the _directories_ the files are going to be created in > with group apache (or whatever group your webserver runs under), with > the sgid bit set, doesn't that cause the file to be created with proper > permission for the cgi? > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk
What are the results using X-lite to X-Lite to X-Lite, does transfer work ? It works when using FWD for us ATA's to and from X-Lite work as well using FWD and other SIP proxies. What are the other endpoints that are being used in the transfer process? Not sure what's going on with the SNOMs or the Budgetones, we have not tested with those devices, I am not even sure if they support transfer, a packet dump would be good, you can send it directly to [EMAIL PROTECTED] Cheers, Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst Sent: Wednesday, July 30, 2003 2:45 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk I have the same with the transfer issue but also when I call between X-Lite and a SNOM 200 there is no audio but if I call between X-Lite and a Budgetone 102 all is OK. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven J. Sobol Sent: 30 July 2003 20:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk On Wed, 30 Jul 2003, Brian West wrote: > Same here. Same build. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
On Wed, 2003-07-30 at 22:33, Patrick wrote: > Did it work after you left a new voice mail message? > > I was looking into the source code to fix it so that the euid was set to > nobody, create the file and then change it back to uid 0, but that didn't > work. Or, maybe change the file mode was 770 with the group set so that > the webserver could modify the file so I wouldn't have to run a suid .cgi > script. If you create the _directories_ the files are going to be created in with group apache (or whatever group your webserver runs under), with the sgid bit set, doesn't that cause the file to be created with proper permission for the cgi? -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] %unsuscribe
On Wed, 2003-07-30 at 22:25, Carlos Crembil wrote:
> %unsuscribe
variable subsitution on the mailinglist contents of asterisk is not
implemented.
If i were, the correct syntax probably would have been:
exten => _asterisk,1,Agi(mailinglist,%{unsubscribe})
There's a link on the bottom of this mail. You'll have better luck
there.
To the list admin, maybe it should say 'to unsubscribe, send mail to
...', just to be more foolproof?
--
'Just when you make something foolproof, the release a better fool.'
Armand.
signature.asc
Description: This is a digitally signed message part
RE: [Asterisk-Users] MGCP behind NAT
My trouble is that the MGCP devices lost the connection with the asterisk My gateways are ASKEY MGCP Any comments? Humberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Miércoles, 30 de Julio de 2003 05:07 p.m. To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] MGCP behind NAT Hello Wade: There is MGCP firmware available too which appears to help in NAT mode. I have been playing around with these CPG for quite a few months now. I have been able to get them to work only with G711 codecs. I was unable to create a coding profile to work successfully with either G723 and G729 codecs. Have you tried to use these codecs on the DG104S ??? Maybe we exchange some notes on this. Ish --__--__-- Message: 8 From: "Wade Weppler" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Wed, 30 Jul 2003 15:09:35 -0400 Subject: [Asterisk-Users] MGCP behind NAT Reply-To: [EMAIL PROTECTED] Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This works: nat=yes line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 This doesn't: line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 nat=yes This makes sense if lines were treated as individual channels through NAT, but they aren't. NAT capability is dictated by the Gateway itself, and not each endpoint/subchannel. I hope this saves somebody some time. -wade ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
I was quoted $75 and $85 USD today. Ricardo Villa http://www.telesip.net - Original Message - From: "Joe Cooke" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:31 PM Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 & 102 > I was quoted the $75 and $85 USD prices from Grandstream direct about 2 > months ago. I'm not sure if it makes a difference, but I live in the US. > > - Joe > - Original Message - > From: "marrandy" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 7:17 PM > Subject: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > > > > > Checking the earlier mails, it stated that the phones were $75 (100) & $85 > > (102) ref :- > > > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html > > > > Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the > person > > said there was no price change. > > > > Anyone on this list actually bought them at the $75 & $85 rate ??? > > > > Regards...Martin > > -- > > Too much is just enough. > > -- Mark Twain, on whiskey > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
I was quoted the $75 and $85 USD prices from Grandstream direct about 2 months ago. I'm not sure if it makes a difference, but I live in the US. - Joe - Original Message - From: "marrandy" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 7:17 PM Subject: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > Checking the earlier mails, it stated that the phones were $75 (100) & $85 > (102) ref :- > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html > > Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person > said there was no price change. > > Anyone on this list actually bought them at the $75 & $85 rate ??? > > Regards...Martin > -- > Too much is just enough. > -- Mark Twain, on whiskey > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
I was quoted $85 per unit, and $80 shipping to New Zealand! - Original Message - From: "marrandy" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 31, 2003 11:17 AM Subject: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > Checking the earlier mails, it stated that the phones were $75 (100) & $85 > (102) ref :- > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html > > Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person > said there was no price change. > > Anyone on this list actually bought them at the $75 & $85 rate ??? > > Regards...Martin > -- > Too much is just enough. > -- Mark Twain, on whiskey > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SCO/Linux concerns
Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users digest, Vol 1 #935 - 14 msgs > Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > >1. RE: voicemail file access problems (Todd Lieberman) >2. sip -> h323 -> ptsn (Brian West) >3. RE: voicemail file access problems (Todd Lieberman) >4. Re: voicemail file access problems (Tilghman Lesher) >5. Re: sip -> h323 -> ptsn (Patrick) >6. RE: voicemail file access problems (Patrick) >7. Re: sip -> h323 -> ptsn (Brian West) >8. Re: sip -> h323 -> ptsn (Patrick) >9. X100P and incoming Context + CDR? (Darren Smith) > 10. Re: CVS Problem? (Kyle Hagan) > 11. Re: sip -> h323 -> ptsn (Eric Wieling) > 12. %unsuscribe (Carlos Crembil) > 13. Re: SetCIDName (Siggi Langauf) > 14. RE: X-Lite and Call transfer using Asterisk (Stuart Hirst) > > --__--__-- > > Message: 1 > From: "Todd Lieberman" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:49:56 -0400 > Reply-To: [EMAIL PROTECTED] > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 2 > Date: Wed, 30 Jul 2003 15:08:53 -0500 (CDT) > From: Brian West <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: [EMAIL PROTECTED] > > I have this setup: > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > Sip phones are setup for out of band dtmf > > but the h323 gateway is inband. Is their a way to pass the digits from > the sip phones to the ptsn via the h323 gateway? > > bkw > > --__--__-- > > Message: 3 > From: "Todd Lieberman" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 16:12:59 -0400 > Reply-To: [EMAIL PROTECTED] > > I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. > > you still need to make sure nobody has read/write permission on > /var/spool/asterisk/vm/$MBOX > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-U
[Asterisk-Users] Grandstream Budgettone 100 & 102
Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough. -- Mark Twain, on whiskey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip -> h323 -> ptsn
Which codec are you using? and which H.323 channel driver? chan_h323 or chan_oh323 ? On 30 Jul 2003, Eric Wieling wrote: > That only works if you are using the G711 (ulaw/alaw) codecs. Other > codecs distort inband DTMF. > > On Wed, 2003-07-30 at 15:26, Patrick wrote: > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > I have this setup: > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > Sip phones are setup for out of band dtmf > > > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > > the sip phones to the ptsn via the h323 gateway? > > > > > > bkw > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip -> h323 -> ptsn
thats all we use right now On Wed, 30 Jul 2003, Eric Wieling wrote: > That only works if you are using the G711 (ulaw/alaw) codecs. Other > codecs distort inband DTMF. > > On Wed, 2003-07-30 at 15:26, Patrick wrote: > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > I have this setup: > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > Sip phones are setup for out of band dtmf > > > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > > the sip phones to the ptsn via the h323 gateway? > > > > > > bkw > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > BTEL Consulting > 850-484-4535 x2111 (Office) > 504-595-3916 x2111 (Experimental) > 877-552-0838 (Backup Phone) > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP behind NAT
Hello Wade: There is MGCP firmware available too which appears to help in NAT mode. I have been playing around with these CPG for quite a few months now. I have been able to get them to work only with G711 codecs. I was unable to create a coding profile to work successfully with either G723 and G729 codecs. Have you tried to use these codecs on the DG104S ??? Maybe we exchange some notes on this. Ish --__--__-- Message: 8 From: "Wade Weppler" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Date: Wed, 30 Jul 2003 15:09:35 -0400 Subject: [Asterisk-Users] MGCP behind NAT Reply-To: [EMAIL PROTECTED] Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This works: nat=yes line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 This doesn't: line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 nat=yes This makes sense if lines were treated as individual channels through NAT, but they aren't. NAT capability is dictated by the Gateway itself, and not each endpoint/subchannel. I hope this saves somebody some time. -wade ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
This is getting too confusing for me ;-( Could someone summarize what are the steps necessary to make vmail.cgi work on a system? Something like this: 1) copy vmail.cgi to your cgi-bin directory 2) copy images/*.gif to your img directory 3) grant 4) grant -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: July 30, 2003 5:33 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Did it work after you left a new voice mail message? I was looking into the source code to fix it so that the euid was set to nobody, create the file and then change it back to uid 0, but that didn't work. Or, maybe change the file mode was 770 with the group set so that the webserver could modify the file so I wouldn't have to run a suid .cgi script. Patrick On Wed, 30 Jul 2003, Todd Lieberman wrote: > I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. > > you still need to make sure nobody has read/write permission on > /var/spool/asterisk/vm/$MBOX > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk
I have the same with the transfer issue but also when I call between X-Lite and a SNOM 200 there is no audio but if I call between X-Lite and a Budgetone 102 all is OK. Stuart -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven J. Sobol Sent: 30 July 2003 20:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk On Wed, 30 Jul 2003, Brian West wrote: > Same here. Same build. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SetCIDName
On Wed, 30 Jul 2003, Jeremy McNamara wrote: > Because H.323 doesn't have a specific 'feature' of caller*id. However, it does seem to have - calling party number - calling party name - display string and at least the last one seems to be set to whatever SetCallerID() tells it to be if you're using chan_oh323 from inaccessnetworks, so that string is displayed on the called party's phone... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] %unsuscribe
%unsuscribe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip -> h323 -> ptsn
That only works if you are using the G711 (ulaw/alaw) codecs. Other codecs distort inband DTMF. On Wed, 2003-07-30 at 15:26, Patrick wrote: > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > On Wed, 30 Jul 2003, Brian West wrote: > > > I have this setup: > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > Sip phones are setup for out of band dtmf > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > the sip phones to the ptsn via the h323 gateway? > > > > bkw > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS Problem?
I figured it out. I had a file called CVS in the directory and it freaked out.. Kyle - Original Message - From: Kyle Hagan To: [EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 9:23 AM Subject: [Asterisk-Users] CVS Problem? Since yesterday i get the following message when downloading anything from the CVS. cvs [checkout aborted]: reading CVS/Tag: Not a directory Is it a problem on my end or digium? I havnt changed anything on my end. Kyle
[Asterisk-Users] X100P and incoming Context + CDR?
Hi folks I have a X100P in my home asterisk box and I have it setup as a default context of 'incoming-pstn' in my extensions.conf i have [incoming-pstn] exten => s,1,Goto(incoming,01225,1) then: [incoming] exten => 01225,1,Answer exten => 01225,2,Dial(SIP/data|m) etc etc Anywho back to the plot. It all works wonderful, someone dials my home office line, asterisk answers, plays them the contents of my mp3 partition whilst my SIP phone rings, I answer and talk to the poor soul about my useless taste in music. However, in the CDR records it says the destination number is 's', is there anyway I can change this? Someone mentioned there was a app_setDNIS function at some point but it seems to have vanished again, or can i do it directly in asterisk/zaptel? Regards Darren Smith Game Digital Ltd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip -> h323 -> ptsn
Try setting dtmf-relay h245-alphanumeric in the MC3810 dial-peer. On Wed, 30 Jul 2003, Brian West wrote: > I have done that but I think its the Asterisk => MC3810 via h323 thats > causing that. Does anyone have an example on how i can dump sip to and > from the MC3810 to my asterisk server? > > bkw > > On Wed, 30 Jul 2003, Patrick wrote: > > > > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > I have this setup: > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > Sip phones are setup for out of band dtmf > > > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > > the sip phones to the ptsn via the h323 gateway? > > > > > > bkw > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip -> h323 -> ptsn
I have done that but I think its the Asterisk => MC3810 via h323 thats causing that. Does anyone have an example on how i can dump sip to and from the MC3810 to my asterisk server? bkw On Wed, 30 Jul 2003, Patrick wrote: > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > On Wed, 30 Jul 2003, Brian West wrote: > > > I have this setup: > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > Sip phones are setup for out of band dtmf > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > the sip phones to the ptsn via the h323 gateway? > > > > bkw > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
Did it work after you left a new voice mail message? I was looking into the source code to fix it so that the euid was set to nobody, create the file and then change it back to uid 0, but that didn't work. Or, maybe change the file mode was 770 with the group set so that the webserver could modify the file so I wouldn't have to run a suid .cgi script. Patrick On Wed, 30 Jul 2003, Todd Lieberman wrote: > I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. > > you still need to make sure nobody has read/write permission on > /var/spool/asterisk/vm/$MBOX > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip -> h323 -> ptsn
I have the same setup, and in the sip.conf file I set the dtmfmode=inband for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. On Wed, 30 Jul 2003, Brian West wrote: > I have this setup: > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > Sip phones are setup for out of band dtmf > > but the h323 gateway is inband. Is their a way to pass the digits from > the sip phones to the ptsn via the h323 gateway? > > bkw > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail file access problems
On Wednesday 30 July 2003 02:49 pm, Todd Lieberman wrote: > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] > Setuid/gid script is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi chmod o-w vmail.cgi btw, 'man chmod' helps. Blindly executing commands as root that you received on a public mailing list is usually not a fine idea. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. you still need to make sure nobody has read/write permission on /var/spool/asterisk/vm/$MBOX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Todd Lieberman Sent: Wednesday, July 30, 2003 3:50 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems I did the chown and now I get [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script is writable by world., referer: http://asterisk.weichertrents.com/cgi-bin/vmail.cgi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paulo Mannheimer Sent: Wednesday, July 30, 2003 3:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > Hi folks, > > I'm having problems accessing my voicemail files through the web > interface. > > I remember that this was discussed on the list, and it seems to be > a permission problem, but I couldn't find any answer by searching > the archives. > > Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip -> h323 -> ptsn
I have this setup: Sip Phones -> Asterisk -> h323 gateway -> ptsn Sip phones are setup for out of band dtmf but the h323 gateway is inband. Is their a way to pass the digits from the sip phones to the ptsn via the h323 gateway? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
I did the chown and now I get [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script is writable by world., referer: http://asterisk.weichertrents.com/cgi-bin/vmail.cgi -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paulo Mannheimer Sent: Wednesday, July 30, 2003 3:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > Hi folks, > > I'm having problems accessing my voicemail files through the web > interface. > > I remember that this was discussed on the list, and it seems to be > a permission problem, but I couldn't find any answer by searching > the archives. > > Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
I did that and now I get -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paulo Mannheimer Sent: Wednesday, July 30, 2003 3:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] voicemail file access problems Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > Hi folks, > > I'm having problems accessing my voicemail files through the web > interface. > > I remember that this was discussed on the list, and it seems to be > a permission problem, but I couldn't find any answer by searching > the archives. > > Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail file access problems
Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: July 30, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicemail file access problems On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > Hi folks, > > I'm having problems accessing my voicemail files through the web > interface. > > I remember that this was discussed on the list, and it seems to be > a permission problem, but I couldn't find any answer by searching > the archives. > > Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk
On Wed, 30 Jul 2003, Brian West wrote: > Same here. Same build. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP behind NAT
Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This works: nat=yes line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 This doesn't: line => aaln/1 line => aaln/2 line => aaln/3 line => aaln/4 nat=yes This makes sense if lines were treated as individual channels through NAT, but they aren't. NAT capability is dictated by the Gateway itself, and not each endpoint/subchannel. I hope this saves somebody some time. -wade ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail file access problems
On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > Hi folks, > > I'm having problems accessing my voicemail files through the web > interface. > > I remember that this was discussed on the list, and it seems to be > a permission problem, but I couldn't find any answer by searching > the archives. > > Any hint? chown root vmail.cgi chmod u+s vmail.cgi -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail file access problems
Hi folks, I’m having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldn’t find any answer by searching the archives. Any hint? PauloHM
RE: [Asterisk-Users] VoiceMail2 Wish List
On Wed, 30 Jul 2003, Benjamin Miller wrote: > I have not had time to complete an "Unified Messaging" component to > voicemail, but I would see this as an admiral goal. Most modern > voicemail systems have some kind of way to delete or mark the voicemail > as read when the message is deleted or read from either telephone or > e-mail. > The biggest hurdle I have come across for this is how does the user > enter their e-mail password into a place where asterisk can use it to > log into a users mail box an actually use it as the sole repository for > mail messages. There's a simple alternative: just setup a master account on the email server that is allowed to access all users' mails and let * use that. Cyrus supports this well. Most "real" IMAP servers should be able to do that... > I see the tasks that need to be completed are: > A) abstract file storage and manipulation in voicemail2 to allow an > "imap" or other type (sql?) of storage plug-in rather than dependency on > a specific file system. Yes, that would be great! > B) an interface to allow the end user to _securly_ enter the username > and password that will be used by asterisk to access the file store. It > needs to be secure so that people who have integrated passwords like > Exchange/AD aren't passing the keys to the kingdom over plain text. An admin account for * would solve that... Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk
Same here. Same build. On Wed, 30 Jul 2003, Dan wrote: > Hi Erik, > > I have the version "X-Lite 2.0 private build 1050". > When I click on transfer then extension then transfer, the call is closed, > but the final destination does not ring. > > Thanks, > Dan > > > - Original Message - > From: "Erik Lagerway" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 8:38 PM > Subject: RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk > > > > Hi Dan, > > > > We had problems with that during a transitional build but the new v2.0 > build > > 1050 should have fixed that, can you confirm that you are using the most > > recent build? > > > > -Erik > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Dan > > Sent: Wednesday, July 30, 2003 9:24 AM > > To: Asterisk Users > > Subject: [Asterisk-Users] X-Lite and Call transfer using Asterisk > > > > > > Hi, > > > > Anyone succeed using call transfer function in X-Lite? > > It is stated that this feature is available in the Lite version too, but > for > > me it doesn't work. > > Clicking on Transfer button, then entering the number and then clicking > > again on transfer doesn't work. > > > > I miss something? > > > > Thanks, > > Dan > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
I believe this is fixed. Sorry. Mark On Wed, 30 Jul 2003 [EMAIL PROTECTED] wrote: > > Hi, > > I am using the latest cvs release of asterisk, and the behaviour is in fact > the same, > > outbound calls work fine, > but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked" > by asterisk, and never reach the phone. > > The setup is the same : 7960 <--> asterisk <--> C2651<-> PSTN > > Yves > > > |-+-> > | | "Low, Adam" | > | | <[EMAIL PROTECTED]>| > | | Sent by: | > | | [EMAIL PROTECTED]| > | | .digium.com | > | | | > | | | > | | 30/07/2003 11:37 | > | | Please respond to | > | | asterisk-users| > | | | > |-+-> > > >---| > | > | > | To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> > | > | cc: > | > | Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 > | > > >---| > > > > > All, > > I've found problems in my setup with the latest couple of revisions > (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 > asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything > is in the same VLAN and only running SIP. > > Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 > > But inbound calls fail, I see the initial INVITE from the AS5300 which is > received by asterisk but not responded to and then the AS5300 sends another > few INVITE's which are received but ignored assumable as they were > duplicates for the first. > > Unfortunately since I've been trying the different cvs revisions of > chan_sip.c I've got susbequent problems with the server crashing after the > first INVITE from the AS5300 using anything greater than cvs 1.134 > > I suspect this is something to do with the per-user limits added in cvs > 1.135 but I am curious to see if anyone has any problems with the latest > cvs elease of asterisk with SIP ? > > Adam > > Sip read: > INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 > Via: SIP/2.0/UDP 213.160.252.50:53893 > From: "611012210" > To: > Date: Wed, 30 Jul 2003 09:26:11 GMT > Call-ID: [EMAIL PROTECTED] > Cisco-Guid: 1667049428-3407675953-0-149543808 > User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled > CSeq: 101 INVITE > Max-Forwards: 6 > Timestamp: 1059557171 > Contact: > Expires: 180 > Content-Type: application/sdp > Content-Length: 149 > > v=0 > o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 > s=SIP Call > c=IN IP4 213.160.252.50 > t=0 0 > m=audio 20032 RTP/AVP 8 0 65535 18 > > 15 headers, 6 lines > Using latest request as basis request > Sending to 213.160.252.50 : 53893 (non-NAT) > Found audio format 8 > Found audio format 0 > Found audio format 65535 > Found audio format 18 > Capabilities: us - 524302, them - 268/0, combined - 12 > Non-codec capabilities: us - 1, them - 0, combined - 0 > AM00CM01*CLI> > Disconnected from Asterisk server > > > * DISCLAIMER * > > This message and any attachment are confidential and may be privileged or > otherwise protected from disclosure and may include proprietary > information. If you are not the intended recipient, please telephone or > email the sender and delete this message and any attachment from your > system. If you are not the intended recipient you must not copy this > message or attachment or disclose the contents to any other person > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk
Hi Erik, I have the version "X-Lite 2.0 private build 1050". When I click on transfer then extension then transfer, the call is closed, but the final destination does not ring. Thanks, Dan - Original Message - From: "Erik Lagerway" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 8:38 PM Subject: RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk > Hi Dan, > > We had problems with that during a transitional build but the new v2.0 build > 1050 should have fixed that, can you confirm that you are using the most > recent build? > > -Erik > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Dan > Sent: Wednesday, July 30, 2003 9:24 AM > To: Asterisk Users > Subject: [Asterisk-Users] X-Lite and Call transfer using Asterisk > > > Hi, > > Anyone succeed using call transfer function in X-Lite? > It is stated that this feature is available in the Lite version too, but for > me it doesn't work. > Clicking on Transfer button, then entering the number and then clicking > again on transfer doesn't work. > > I miss something? > > Thanks, > Dan > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk
Hi Dan, We had problems with that during a transitional build but the new v2.0 build 1050 should have fixed that, can you confirm that you are using the most recent build? -Erik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Sent: Wednesday, July 30, 2003 9:24 AM To: Asterisk Users Subject: [Asterisk-Users] X-Lite and Call transfer using Asterisk Hi, Anyone succeed using call transfer function in X-Lite? It is stated that this feature is available in the Lite version too, but for me it doesn't work. Clicking on Transfer button, then entering the number and then clicking again on transfer doesn't work. I miss something? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail2 unified messaging
We have built this in-house and it works well but we had to overcome some
challenges along the way. For example,
1) Your voicemail server may be off-net (behind NAT or using private ip
addresses) while your email users are on-net. The way we approached this was
to abstract the functionality such that an always on-net script accepts the
request and then calls a cgi on the off-net server.
2) The link in email needs VM_CONTEXT because that's the directory the
voicemail is stored in. The current source code doesn't pass this into the
sendmail function so we added this for ourselves but didn't request it added
to cvs because we weren't sure it would be valuable for others.
3) The email link also needs a unique voicemail ID so it knows which one to
delete. However, in the current voicemail implementation there is no such
thing. MsgID certainly isn't it because as soon as you delete msg then
the messages get reordered and msg0001 becomes msg, et cetera. So if you
delete msg and then reorder three times in a row, you are literally
deleting three distinct messages. The best we've found - which isn't really
all that good - is a variable in msg.txt called origtime. We hacked
voicemail.c to pass this variable to the sendmail function too.
So our implementation goes something like this...
in voicemail.conf we have
emailbody=Dear ${VM_NAME},\n\nYou were just left a ${VM_DUR} long message
(number ${VM_MSGNUM}) in mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on
${VM_DATE}. You can listen to the message by clicking on the attached
file.\n\nOnce you listen to the attached sound file, you may optionally
delete the message from your voicemail box on the phone system by clicking
here:\n\nTO DELETE:
http://www.abc.com/login/rmvm.html?mailbox=${VM_MAILBOX}&origtime=${VM_START
}&context=${VM_CONTEXT}.
As this is a password-protected page within our intranet, we now do a
database query to ensure that the person currently logged in actually owns
the mailbox passed in by the link. This protects us from one user hacking
the http get string to delete someone else's email.
At this point our rmvm.html script executes a DeleteVoicemail script (via
ssh) on our off-net asterisk server. The Delete Voicemail script greps
directory /var/lib/asterisk/sounds/voicemail/VM_CONTEXT/VM_MAILBOX for
VM_START (origtime) and finds the MsgID to delete (let's say msg0001). Then
it does an rm on msg0001.* and finally reorders all of the messages so
msg0002.* becomes msg0001.*, msg0003.* becomes msg0002.*, et cetera.
This system is pretty convoluted but works great for us. It's fun to see our
vm indicator light come on, get an email, click on a link in the email body,
and then watch the vm indicator light go off without ever calling into
voicemail. I just don't think a universal solution is going to help much
unless everybody puts their asterisk servers on-net and you don't want
things like intranet user authentication.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Todd
> Lieberman
> Sent: Wednesday, July 30, 2003 10:09 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] VoiceMail2 Wish List
>
>
> Will there be a way to delete messages from email. I love
> getting voicemail
> in wav to my email, but I hate having to delete them when I call in to get
> my messages. If we could add a link and have a cgi delete the
> messages that
> would be a nice time saver. TL
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Brad Bergman
> Sent: Wednesday, July 30, 2003 4:18 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] VoiceMail2 Wish List
>
>
> On July 29, 2003 01:57 am, Roy Sigurd Karlsbakk wrote:
> > > Ah, now that you mention it, I implemented this in my patch also and
> then
> > > forgot about it: messages that are too short (less than 3 seconds) or
> all
> > > silence
> >
> > Perhaps this should be configurable?
>
> Yeah, I suppose it should. I added minlength (in seconds) and removesilent
> (yes/no) as [general] options, with 3 and yes as defaults, respectively.
>
> Brad
> ___
> Asterisk-Users mailing list
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>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
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>
___
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Re: [Asterisk-Users] Some stats
On Wed, 2003-07-30 at 09:40, Rattana BIV wrote: > Hi, > > I try to make some statistics about call on Asterisk. Is there > something who makes it ? > I will be interesting to have the time of a call and a list of current > calls. Current calls can be found either from the CLI, or from the manager interface. Completed calls are all in the /var/log/asterisk/cdr-csv/Master.csv -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail2 Wish List
I'm not running a high security solution so an link in w/a MD5 encrypted path name or guid would be sufficient. I don't want to enter a password every time. Can Voicemail2 save the files by unique file name? I'd be happy to write the cgi that deletes the message or marks it as read. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Benjamin Miller Sent: Wednesday, July 30, 2003 12:27 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoiceMail2 Wish List Actually, this is a much bigger task than you would imagine. I have not had time to complete an "Unified Messaging" component to voicemail, but I would see this as an admiral goal. Most modern voicemail systems have some kind of way to delete or mark the voicemail as read when the message is deleted or read from either telephone or e-mail. The biggest hurdle I have come across for this is how does the user enter their e-mail password into a place where asterisk can use it to log into a users mail box an actually use it as the sole repository for mail messages. I see the tasks that need to be completed are: A) abstract file storage and manipulation in voicemail2 to allow an "imap" or other type (sql?) of storage plug-in rather than dependency on a specific file system. B) an interface to allow the end user to _securly_ enter the username and password that will be used by asterisk to access the file store. It needs to be secure so that people who have integrated passwords like Exchange/AD aren't passing the keys to the kingdom over plain text. Just my 2 cents worth. Ben -Original Message- From: Todd Lieberman [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoiceMail2 Wish List Will there be a way to delete messages from email. I love getting voicemail in wav to my email, but I hate having to delete them when I call in to get my messages. If we could add a link and have a cgi delete the messages that would be a nice time saver. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brad Bergman Sent: Wednesday, July 30, 2003 4:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMail2 Wish List On July 29, 2003 01:57 am, Roy Sigurd Karlsbakk wrote: > > Ah, now that you mention it, I implemented this in my patch also and then > > forgot about it: messages that are too short (less than 3 seconds) > > or all > > silence > > Perhaps this should be configurable? Yeah, I suppose it should. I added minlength (in seconds) and removesilent (yes/no) as [general] options, with 3 and yes as defaults, respectively. Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail2 Wish List
Actually, this is a much bigger task than you would imagine. I have not had time to complete an "Unified Messaging" component to voicemail, but I would see this as an admiral goal. Most modern voicemail systems have some kind of way to delete or mark the voicemail as read when the message is deleted or read from either telephone or e-mail. The biggest hurdle I have come across for this is how does the user enter their e-mail password into a place where asterisk can use it to log into a users mail box an actually use it as the sole repository for mail messages. I see the tasks that need to be completed are: A) abstract file storage and manipulation in voicemail2 to allow an "imap" or other type (sql?) of storage plug-in rather than dependency on a specific file system. B) an interface to allow the end user to _securly_ enter the username and password that will be used by asterisk to access the file store. It needs to be secure so that people who have integrated passwords like Exchange/AD aren't passing the keys to the kingdom over plain text. Just my 2 cents worth. Ben -Original Message- From: Todd Lieberman [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 11:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoiceMail2 Wish List Will there be a way to delete messages from email. I love getting voicemail in wav to my email, but I hate having to delete them when I call in to get my messages. If we could add a link and have a cgi delete the messages that would be a nice time saver. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brad Bergman Sent: Wednesday, July 30, 2003 4:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMail2 Wish List On July 29, 2003 01:57 am, Roy Sigurd Karlsbakk wrote: > > Ah, now that you mention it, I implemented this in my patch also and then > > forgot about it: messages that are too short (less than 3 seconds) > > or all > > silence > > Perhaps this should be configurable? Yeah, I suppose it should. I added minlength (in seconds) and removesilent (yes/no) as [general] options, with 3 and yes as defaults, respectively. Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite and Call transfer using Asterisk
Hi, Anyone succeed using call transfer function in X-Lite? It is stated that this feature is available in the Lite version too, but for me it doesn't work. Clicking on Transfer button, then entering the number and then clicking again on transfer doesn't work. I miss something? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS Problem?
Since yesterday i get the following message when downloading anything from the CVS. cvs [checkout aborted]: reading CVS/Tag: Not a directory Is it a problem on my end or digium? I havnt changed anything on my end. Kyle
RE: [Asterisk-Users] Microsoft SQL: cdr_tds.c
We have talked about a cdr_tds.c how many times now on this list? I recently rewrote cdr_mysql.c to use a new table structure CREATE TABLE cdr ( accountcode varchar(45) NOT NULL default '', src varchar(45) NOT NULL default '', dst varchar(45) NOT NULL default '', dcontext varchar(45) NOT NULL default '', clid varchar(45) NOT NULL default '', channel varchar(45) NOT NULL default '', dstchannel varchar(45) NOT NULL default '', lastapp varchar(45) NOT NULL default '', lastdata varchar(45) NOT NULL default '', starttime datetime NOT NULL default '-00-00 00:00:00', answertime datetime NOT NULL default '-00-00 00:00:00', endtime datetime NOT NULL default '-00-00 00:00:00', duration int(11) NOT NULL default '0', billsec int(11) NOT NULL default '0', disposition varchar(45) NOT NULL default '', amaflags varchar(45) NOT NULL default '', KEY src (src), KEY clid (clid), KEY dst (dst), KEY accountcode (accountcode) ) COMMENT='Asterisk CDR table'; The old cdr_mysql.c interface did not track enough CDR data for tracking calls. I also wrote a lot of PHP pages to get metrics on this information from this new table. I am not sure it is useful to others because we only wanted to track VOIP SIP/H323 meetme conferencing usage. We do not have PSTN lines into Asterisk so it is in no way a billing interface. FreeTDS is a great and stable library. I use it to interface between my Linux code and my VMWare running MS SQL Server 2000. If there is a demand (and a couple hundred $ to cover 3 days of side work) I would be more than happy to write a cdr_tds.c/readme_tds.txt and email/commit it to asterisk. I have to charge a little to cover my time. Sorry, we are using MySQL, not using MS SQL or Sybase, for my "meetme billing" information. If we were using a TDS Database I would have written a FreeTDS interface LONG ago for my company open sourced/released it to the community. If someone needs such a feature let me know. I will be more than happy to oblige. Erik > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven J. > Sobol > Sent: Wednesday, July 30, 2003 9:48 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Microsoft SQL > > > On Wed, 30 Jul 2003, Florian Overkamp wrote: > > > As suggested by another poster: MS-SQL is mostly based on > Sybase, so any > > Sybase driver (there is one for PHP for instance) can probably be used, > > from AGI or otherwise... > > I've successfully used the FreeTDS libraries on a Linux box to connect to > a MySQL server. The original MS SQL was based on Sybase SQL Server 4.2, > but for SQL 7 or SQL 2000, you'll want to compile the FreeTDS library with > version 7.0 support (you can either do 7.0 or 4.2). > > -- > JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & > Connectivity] > 22674 Motnocab Road * Apple Valley, CA 92307-1950 > Steve Sobol, Proprietor > 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail2 Wish List
Will there be a way to delete messages from email. I love getting voicemail in wav to my email, but I hate having to delete them when I call in to get my messages. If we could add a link and have a cgi delete the messages that would be a nice time saver. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brad Bergman Sent: Wednesday, July 30, 2003 4:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VoiceMail2 Wish List On July 29, 2003 01:57 am, Roy Sigurd Karlsbakk wrote: > > Ah, now that you mention it, I implemented this in my patch also and then > > forgot about it: messages that are too short (less than 3 seconds) or all > > silence > > Perhaps this should be configurable? Yeah, I suppose it should. I added minlength (in seconds) and removesilent (yes/no) as [general] options, with 3 and yes as defaults, respectively. Brad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk,ata186 and Panasonic TD1232
I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk. Can I dial from asterisk into ata, then indicate phone number playing tone (use DISA feature at panasonic) and connect to any analog phone connected to panasonic ? I think some of Playtones application within Dial application can help me. But I don't know how. -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audiocodes fxs
Try ftp://angelgomez.homelinux.com/pub/audiocodes Anton Tinchev wrote: Kelvin Chua wrote: hi guys, have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing? ~kelvin Can someone send me SIP firmwire for audiocodes 104. I has h.323 only and it sucks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rxgain and txgain in zapata.conf
Hi, Do you have some experience with the "best" values for those parameters in youyr particular case? I mean the best raport between sound level in both direction and echo cancellation. For me, the best result I can get is with: rxgain=10 txgain=15 ... the sound level is good, but the echo is a little bit to strong for my taste. Something interesting is that if I put a txgain value greater than 15 (let's say 20), the external dial through X100P is not corrrect anymore. I get from the PSTN operator the message " the number you have dialed is invalid". Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dummy account/extension
Hi, It was one of the possible way to get a workaround with my ATA and attended call transfer. It was solved for now.. check one of my previous mails. Thanks, Dan - Original Message - From: "Armand A. Verstappen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:11 PM Subject: Re: [Asterisk-Users] Dummy account/extension ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail message forwarded to another extension and file format changing
Thanks Ben. I have entered an item in the bug track on Digium site. This is not a real issue for me, more an observation. It can be an issue in some circumstances, but I'm sure that it will be solved in the near future. Thank you again for your info. Dan - Original Message - From: "Benjamin Miller" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:17 PM Subject: RE: [Asterisk-Users] Voicemail message forwarded to another extension and file format changing > Check the source on this. When I send Mark a patch way back when to > allow the e-mail to be triggered to the person who gets the forwarded > message, I selfishly only coded it to handle a wav file. Mark may have > fixed or changed this at a later point, but if not, there is probably a > to-do comment in the code to fix this. > Ben > > -Original Message- > From: Dan [mailto:[EMAIL PROTECTED] > Sent: Wednesday, July 30, 2003 3:30 AM > To: Asterisk Users > Subject: [Asterisk-Users] Voicemail message forwarded to another > extension and file format changing > > > Hi, > > I have configured voicemail to use GSM as the file format and to send > the voice message to an e-mail address. It works, and the destination > receive the voice message attached in the same GSM format. When I try to > forward the voice message from my voicemail to another extension, the > voice message is forwarded, but the destination receive the e-mail with > the voice message in WAV format (the Asterisk type of WAV), even it is > the same one. This is a bug or I miss something? > > There is any way to select a different file format for the forwarded > message? > > Thanks, > Dan > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk installation
On Wednesday 30 July 2003 08:06 am, Wen Wen wrote: > great! it works. I am now able to get the cli prompt with "asterisk > -vvvcg" > > when i try to use "/usr/sbin/safe_asterisk" script to start, i > still got the error of > "Asterisk ended with exit status 127". I guess I can use the > "asterisk -vvvcg" to > start the server. but I would like to know why the "safe_asterisk" > does not work. It might be because safe_asterisk is attempting to write to a TTY which does not exist. It's just a shell script, so you can easily edit it to use a TTY on the console which does exist. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail message forwarded to another extension and file format changing
Check the source on this. When I send Mark a patch way back when to allow the e-mail to be triggered to the person who gets the forwarded message, I selfishly only coded it to handle a wav file. Mark may have fixed or changed this at a later point, but if not, there is probably a to-do comment in the code to fix this. Ben -Original Message- From: Dan [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 30, 2003 3:30 AM To: Asterisk Users Subject: [Asterisk-Users] Voicemail message forwarded to another extension and file format changing Hi, I have configured voicemail to use GSM as the file format and to send the voice message to an e-mail address. It works, and the destination receive the voice message attached in the same GSM format. When I try to forward the voice message from my voicemail to another extension, the voice message is forwarded, but the destination receive the e-mail with the voice message in WAV format (the Asterisk type of WAV), even it is the same one. This is a bug or I miss something? There is any way to select a different file format for the forwarded message? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dummy account/extension
On Wed, 2003-07-30 at 16:44, Dan wrote: > Thanks for the suggestion. > I have change it like that: > > ;dummy extension > exten => 199,1,Ringing > exten => 199,2,Wait(60) ; give illusion we might pick up > exten => 199,3,Hangup > > in order to hear the ring too. > > ..but now... how can I do to call this extension from a Dial command? Not sure what you are trying to do, but would the goto app be of any help? [other-ext] ... exten => 198,3,Goto(dummy,199,1) wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
I also had the same problem with sip, I also moved back a couple of weeks in cvs. I also use a AS5300 Cisco in my call chain. I got a bunch of "Ignoring this request" in debug. I have not had time to trace the call path on this problem yet. Low, Adam wrote: All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: "611012210" To: Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADSI and SoftKeys
On Wed, 2003-07-30 at 16:40, John Congdon wrote: > Has anyone solved the problem on the ADSI phones > that when you hit one of the soft keys, the Number Pad > stops working? No, I haven't. Just confirming that I have the same problem here, using the VoiceMail2 app. Do you experience this outside VoiceMail2 as well? wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] isdn4linux/Teles16.3
Hi, On Wed, 2003-07-30 at 16:15, [EMAIL PROTECTED] wrote: > is it possible to use a Teles16.3 via isdn4linux for the external phone > connections (phone provider net)? Yes, it is. I tested using an old card I had lying around. I quickly switched to a Fritz card and chan_capi however. This solved the big issue I had with echo for me. Since I had both available, I did not spend much time trying to solve the problems I had. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
[Asterisk-Users] Need help
I do part time consulting work. I need to setup an asterisk system to allow me to record both inbound and outbound calls to clients. I have one client that is just a PITA. The client has changed their mind three times so far and we are at step one. I have a spare slackware box and a seperate phone line for the consulting work. I have MCI Neighorhood as my carrier. What I need to know is: 1. What hardware is needed to record both inbound and outbound call with the disclaimer added to the phone call. The hardward should be as transpartent as possible. If I can get this to work I might setup it up on the main line also but with out the recording capibilites. 2. Any special software patchs to apply to the base source. 3. Any tips on setting up the config files. For the recording and blocking numbers that do not have proper caller I like telemarketers. The blocking should be done before the call is passed thur. 4. Need to go as cheep as posible but still have the needed capibilites. 5. I know nothing about telcom except that I pick up the phone, get a dial tone, and dial. If any of the above can or cannot be done please let me know of any alternative, if it possible. Thank you, Donn -- [EMAIL PROTECTED] (remove no spam) Virgina Resident/4530 [EMAIL PROTECTED] WU ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dummy account/extension - Workaround for attended call trabsfer to ATA186
Hi again, I think I have now a workaround for call transfer on ATA 186. This is the extension corresponding to the phone connected to an ATA186 exten => 103,1,Dial(SIP/103,20),Tt exten => 103,2,Voicemail2(us101) exten => 103,3,Hangup exten => 103,102,Ringing exten => 103,103,Wait(1) exten => 103,104,Goto(1) I can now to attended transfer a call to this phone too. The strange thins is that if I call this extension when the phone in off-hook but not in a call, it rings for 1 second then exit with a busy tone. Why? Thanks, Dan - Original Message - From: "Dan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 5:44 PM Subject: Re: [Asterisk-Users] Dummy account/extension > Hi, > > Thanks for the suggestion. > I have change it like that: > > ;dummy extension > exten => 199,1,Ringing > exten => 199,2,Wait(60) ; give illusion we might pick up > exten => 199,3,Hangup > > in order to hear the ring too. > > ..but now... how can I do to call this extension from a Dial command? > > What I want in the final is to have a workaround for ATA186 in order to > prevent consider it busy during the attended transfer. > More, I want to prevent been bussy when not in a call. The Call Waiting does > not function during the dialtone period, just during the call. > > There is any other way to do it? > > Thanks for your help, > Dan > > > > > - Original Message - > From: "Armand A. Verstappen" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 5:06 PM > Subject: Re: [Asterisk-Users] Dummy account/extension > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Microsoft SQL
On Wed, 30 Jul 2003, Steven J. Sobol wrote: > I've successfully used the FreeTDS libraries on a Linux box to connect to > a MySQL server s/MySQL/MS SQL/g -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Microsoft SQL
On Wed, 30 Jul 2003, Florian Overkamp wrote: > As suggested by another poster: MS-SQL is mostly based on Sybase, so any > Sybase driver (there is one for PHP for instance) can probably be used, > from AGI or otherwise... I've successfully used the FreeTDS libraries on a Linux box to connect to a MySQL server. The original MS SQL was based on Sybase SQL Server 4.2, but for SQL 7 or SQL 2000, you'll want to compile the FreeTDS library with version 7.0 support (you can either do 7.0 or 4.2). -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dummy account/extension
Hi, Thanks for the suggestion. I have change it like that: ;dummy extension exten => 199,1,Ringing exten => 199,2,Wait(60) ; give illusion we might pick up exten => 199,3,Hangup in order to hear the ring too. ..but now... how can I do to call this extension from a Dial command? What I want in the final is to have a workaround for ATA186 in order to prevent consider it busy during the attended transfer. More, I want to prevent been bussy when not in a call. The Call Waiting does not function during the dialtone period, just during the call. There is any other way to do it? Thanks for your help, Dan - Original Message - From: "Armand A. Verstappen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 5:06 PM Subject: Re: [Asterisk-Users] Dummy account/extension ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI and SoftKeys
Has anyone solved the problem on the ADSI phones that when you hit one of the soft keys, the Number Pad stops working? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some stats
Hi, I try to make some statistics about call on Asterisk. Is there something who makes it ? I will be interesting to have the time of a call and a list of current calls. Regards Rattana
[Asterisk-Users] isdn4linux/Teles16.3
Hi, is it possible to use a Teles16.3 via isdn4linux for the external phone connections (phone provider net)? Internally I want to use the TDM30B card to connect my analogue interfaces. Lars ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
That also worked for me. My AudioCodes MP-104 FXO has no problem
making inbound calls now.
Thanks Patrick and Adam.
-Brenton
- Original Message -
From: "Low, Adam" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 8:45 AM
Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
> Well found Patrick, that did the trick for me as well !
>
> I had been trying to debug 1.135 where this portion of code wasn't added
yet ... thats a lesson learnt ...
>
> -Original Message-
> From: Patrick
> To: '[EMAIL PROTECTED] '
> Sent: 30/07/03 15:04
> Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
>
>
> It is in the find_user() routine. If it is not an extension on the
> PBX,
> it should return a zero
>
> if ( isfound ) {
>ast_log(LOG_DEBUG, "%s is not a local user\n", name);
>ast_pthread_mutex_unlock(&userl.lock);
>return 1; <--- this is the problem - change it to a 0.
> }
>
> It isn't an error, so it should just return. Change that and the
> function
> will work properly. I tested it using an AS5350 and successly made an
> inbound call.
>
> Patrick
>
>
> On Wed, 30 Jul 2003, Low, Adam wrote:
>
> > Brenton, Yves, ...
> >
> > I've located the cause of the problem in chan_sip.c but am still
> trying to find the exact cause being completely new to the asterisk
> code. It seems that there was an added function in 1.135 called
> 'find_user' that is supposed to lookup the users incoming call limit but
> the routine is unable to find a matching user for my AS5300 which I
> suspect is because it does not REGISTER with the server prior to
> attempting to send calls.
> >
> > I'm going to continue debugging a little later and see if I can narrow
> it down more ...
> >
> > Adam
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > To: [EMAIL PROTECTED]
> > Sent: 30/07/03 14:09
> > Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs
> 1.134
> >
> >
> > Hi,
> >
> > I am using the latest cvs release of asterisk, and the behaviour is in
> > fact
> > the same,
> >
> > outbound calls work fine,
> > but for inbound calls (from C2651 over PSTN) , SIP messages get
> > "blocked"
> > by asterisk, and never reach the phone.
> >
> > The setup is the same : 7960 <--> asterisk <--> C2651<->
> > PSTN
> >
> > Yves
> >
> >
> > |-+->
> > | | "Low, Adam" |
> > | | <[EMAIL PROTECTED]>|
> > | | Sent by: |
> > | | [EMAIL PROTECTED]|
> > | | .digium.com |
> > | | |
> > | | |
> > | | 30/07/2003 11:37 |
> > | | Please respond to |
> > | | asterisk-users|
> > | | |
> > |-+->
> >
> >
> >---
> > |
> > |
> > |
> > | To: "'[EMAIL PROTECTED]'"
> > <[EMAIL PROTECTED]> |
> > | cc:
> > |
> > | Subject: [Asterisk-Users] chan_sip.c problems problems from
> > cvs 1.134 |
> >
> >
> >---
> > |
> >
> >
> >
> >
> > All,
> >
> > I've found problems in my setup with the latest couple of revisions
> > (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9
> > asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's,
> > everything
> > is in the same VLAN and only running SIP.
> >
> > Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300
> >
> > But inbound calls fail, I see the initial INVITE from the AS5300 which
> > is
> > received by asterisk but not responded to and then the AS5300 sends
> > another
> > few INVITE's which are received but ignored assumable as they were
> > duplicates for the first.
> >
> > Unfortunately since I've been trying the different cvs revisions of
> > chan_sip.c I've got susbequent problems with the server crashing after
> > the
> > first INVITE from the AS5300 using anything greater than cvs 1.134
> >
> > I suspect this is something to do with the per-user limits added in
> cvs
> > 1.135 but I am curious to see if anyone has any problems with the
> latest
> > cvs elease of asterisk with SIP ?
> >
> > Adam
> >
> > Sip read:
> > INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0
> > Via: SIP/2.0/UDP 213.160.252.50:53893
> > From: "611012210"
> > To:
> > Date: Wed, 30 Jul 2003 09:26:11 GMT
> > Call-ID: [EMAIL PROTECTED]
> > Cisco-Guid: 1667049428-3407675953-0-149543808
> > User-Agent: Cisco V
Re: [Asterisk-Users] Dummy account/extension
On Wed, 2003-07-30 at 15:55, Dan wrote:
> It is possible to create a dummy account (SIP or IAX type) in order to be
> used in a "dummy" extension?
> I want to be able to use it as a normal extension (as an IP phone connected
> to it), but without the need to answer or call from that extension.
> I want that when I call that extension to hear the ring, and after the
> defined period of time to enter in the Voicemail system.
> I don't want to use a real phone (hardware or software) for this purpose.
>
> It is possible to do this in a simple way?
doesn't:
[globals]
WAITTIME=10
MAILBOX=1234
[dummy]
exten => 1234,1,Wait(${WAITTIME}) ; give illusion we might pick up
exten => 1234,2,VoiceMail2(${MAILBOX}) ; then kick into voicemail
exten => 1234,3,Hangup
do the trick?
wkr,
--
Envida http://www.envida.net/
Armand A. Verstappen Graadt van Roggenweg 328
[EMAIL PROTECTED] 3531 AH Utrecht
tel: +31 (0)30 298 2255Postbus 19127
fax: +31 (0)30 298 21113501 DC Utrecht
signature.asc
Description: This is a digitally signed message part
[Asterisk-Users] Dummy account/extension
Hi, It is possible to create a dummy account (SIP or IAX type) in order to be used in a "dummy" extension? I want to be able to use it as a normal extension (as an IP phone connected to it), but without the need to answer or call from that extension. I want that when I call that extension to hear the ring, and after the defined period of time to enter in the Voicemail system. I don't want to use a real phone (hardware or software) for this purpose. It is possible to do this in a simple way? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicetronix Hardware
///shameless advert/// I have just listed my two 4 channel voicetronix cards on ebay if anyone is interested. The cards work great but we have now moved to E1 so can no longer use analogue cards so have moved over to asterisk running on digium hardware. Items no 2745081372 and 2745081930 I'm not sure of the current state of the voicetronix driver for asterisk but these cards were from my previous attempts at IVR using bayonne. Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Well found Patrick, that did the trick for me as well !
I had been trying to debug 1.135 where this portion of code wasn't added yet ... thats
a lesson learnt ...
-Original Message-
From: Patrick
To: '[EMAIL PROTECTED] '
Sent: 30/07/03 15:04
Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
It is in the find_user() routine. If it is not an extension on the
PBX,
it should return a zero
if ( isfound ) {
ast_log(LOG_DEBUG, "%s is not a local user\n", name);
ast_pthread_mutex_unlock(&userl.lock);
return 1; <--- this is the problem - change it to a 0.
}
It isn't an error, so it should just return. Change that and the
function
will work properly. I tested it using an AS5350 and successly made an
inbound call.
Patrick
On Wed, 30 Jul 2003, Low, Adam wrote:
> Brenton, Yves, ...
>
> I've located the cause of the problem in chan_sip.c but am still
trying to find the exact cause being completely new to the asterisk
code. It seems that there was an added function in 1.135 called
'find_user' that is supposed to lookup the users incoming call limit but
the routine is unable to find a matching user for my AS5300 which I
suspect is because it does not REGISTER with the server prior to
attempting to send calls.
>
> I'm going to continue debugging a little later and see if I can narrow
it down more ...
>
> Adam
>
> -Original Message-
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> Sent: 30/07/03 14:09
> Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs
1.134
>
>
> Hi,
>
> I am using the latest cvs release of asterisk, and the behaviour is in
> fact
> the same,
>
> outbound calls work fine,
> but for inbound calls (from C2651 over PSTN) , SIP messages get
> "blocked"
> by asterisk, and never reach the phone.
>
> The setup is the same : 7960 <--> asterisk <--> C2651<->
> PSTN
>
> Yves
>
>
> |-+->
> | | "Low, Adam" |
> | | <[EMAIL PROTECTED]>|
> | | Sent by: |
> | | [EMAIL PROTECTED]|
> | | .digium.com |
> | | |
> | | |
> | | 30/07/2003 11:37 |
> | | Please respond to |
> | | asterisk-users|
> | | |
> |-+->
>
>
>---
> |
> |
> |
> | To: "'[EMAIL PROTECTED]'"
> <[EMAIL PROTECTED]> |
> | cc:
> |
> | Subject: [Asterisk-Users] chan_sip.c problems problems from
> cvs 1.134 |
>
>
>---
> |
>
>
>
>
> All,
>
> I've found problems in my setup with the latest couple of revisions
> (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9
> asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's,
> everything
> is in the same VLAN and only running SIP.
>
> Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300
>
> But inbound calls fail, I see the initial INVITE from the AS5300 which
> is
> received by asterisk but not responded to and then the AS5300 sends
> another
> few INVITE's which are received but ignored assumable as they were
> duplicates for the first.
>
> Unfortunately since I've been trying the different cvs revisions of
> chan_sip.c I've got susbequent problems with the server crashing after
> the
> first INVITE from the AS5300 using anything greater than cvs 1.134
>
> I suspect this is something to do with the per-user limits added in
cvs
> 1.135 but I am curious to see if anyone has any problems with the
latest
> cvs elease of asterisk with SIP ?
>
> Adam
>
> Sip read:
> INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0
> Via: SIP/2.0/UDP 213.160.252.50:53893
> From: "611012210"
> To:
> Date: Wed, 30 Jul 2003 09:26:11 GMT
> Call-ID: [EMAIL PROTECTED]
> Cisco-Guid: 1667049428-3407675953-0-149543808
> User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
> CSeq: 101 INVITE
> Max-Forwards: 6
> Timestamp: 1059557171
> Contact:
> Expires: 180
> Content-Type: application/sdp
> Content-Length: 149
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50
> s=SIP Call
> c=IN IP4 213.160.252.50
> t=0 0
> m=audio 20032 RTP/AVP 8 0 65535 18
>
> 15 headers, 6 lines
> Using latest request as basis request
> Sending to 213.160.252.50 : 53893 (non-NAT)
> Found audio format 8
> Found audio format 0
> Found audio format 65535
> Found audio format 18
> Capabilities: us - 524302, them - 26
[Asterisk-Users] X100P call detection
How the X100P card detect the arrival of a call? I have a bunch of cards and they perform perfectly when connected to a PBX, but when I connect to another brand of PBX, they don't unhook the line, or in other words, they don't answer the line. If I connect a normal analog phone on the same line (even via the second RJ connector on the card), I can hear the phone ringing! Is there some adjustament to be made to the configuration to make the X100P more sensible to the phone ring? I notice that the old X100P cards, the one without the epox on the chip, were better in detecting the ringing. Leandro
Re: [Asterisk-Users] Call Transfer, Budgettone 100
I was told in #asterisk that you just hit transfer, dial the extension, speak to caller and press transfer once your done talking and it should do it. In addition you can do transfer+extension+transfer+hangup... Thats how I was told it would work. bkw On Wed, 30 Jul 2003, denon wrote: > Last I checked, SIP transfer to park doesn't work .. only way to do it is > using T and a # transfer .. which is ugly. Has this been fixed? > > -d > > At 10:51 AM 7/30/2003 +0200, you wrote: > >park the call > > > >On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote: > > > hi, > > > can someone who has used Budgettone phones tell me how to do the > > > following: > > > > > > an incoming call comes in and is answered by the receptionist. > > > she need to put the call on hold, speak to whoever the call is for, > > > and either (after that) pass on the call, otherwise speak again to > > > whoever was on the call and hang up .. > > > > > > so far i've got as far as a blind transfer by pressing transfer button > > > and then the new extension .. > > > > > > cheers > > > Dave > > > --- > > > Email sent using AnyEmail (http://netbula.com/anyemail/) > > > Netbula LLC is not responsible for the content of this email > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk installation
great! it works. I am now able to get the cli prompt with "asterisk -vvvcg" when i try to use "/usr/sbin/safe_asterisk" script to start, i still got the error of "Asterisk ended with exit status 127". I guess I can use the "asterisk -vvvcg" to start the server. but I would like to know why the "safe_asterisk" does not work. thanx. Wen From: Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> To: [EMAIL PROTECTED],"Wen Wen" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Asterisk installation Date: Wed, 30 Jul 2003 02:03:22 +0200 On Wednesday 30 July 2003 00:29, Wen Wen wrote: > ERROR[1024]: File chan_modem.c, Line 852 (load_module): Unable to load > config modem.conf er add this line to modules.conf noload => chan_modem.so potentially also these noload => chan_modem_aopen.so noload => chan_modem_bestdata.so noload => chan_modem_i4l.so roy (sorry for the repost - sent that last message a little to early) _ Add photos to your e-mail with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
It is in the find_user() routine. If it is not an extension on the PBX,
it should return a zero
if ( isfound ) {
ast_log(LOG_DEBUG, "%s is not a local user\n", name);
ast_pthread_mutex_unlock(&userl.lock);
return 1; <--- this is the problem - change it to a 0.
}
It isn't an error, so it should just return. Change that and the function
will work properly. I tested it using an AS5350 and successly made an
inbound call.
Patrick
On Wed, 30 Jul 2003, Low, Adam wrote:
> Brenton, Yves, ...
>
> I've located the cause of the problem in chan_sip.c but am still trying to find the
> exact cause being completely new to the asterisk code. It seems that there was an
> added function in 1.135 called 'find_user' that is supposed to lookup the users
> incoming call limit but the routine is unable to find a matching user for my AS5300
> which I suspect is because it does not REGISTER with the server prior to attempting
> to send calls.
>
> I'm going to continue debugging a little later and see if I can narrow it down more
> ...
>
> Adam
>
> -Original Message-
> From: [EMAIL PROTECTED]
> To: [EMAIL PROTECTED]
> Sent: 30/07/03 14:09
> Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
>
>
> Hi,
>
> I am using the latest cvs release of asterisk, and the behaviour is in
> fact
> the same,
>
> outbound calls work fine,
> but for inbound calls (from C2651 over PSTN) , SIP messages get
> "blocked"
> by asterisk, and never reach the phone.
>
> The setup is the same : 7960 <--> asterisk <--> C2651<->
> PSTN
>
> Yves
>
>
> |-+->
> | | "Low, Adam" |
> | | <[EMAIL PROTECTED]>|
> | | Sent by: |
> | | [EMAIL PROTECTED]|
> | | .digium.com |
> | | |
> | | |
> | | 30/07/2003 11:37 |
> | | Please respond to |
> | | asterisk-users|
> | | |
> |-+->
>
> >---
> |
> |
> |
> | To: "'[EMAIL PROTECTED]'"
> <[EMAIL PROTECTED]> |
> | cc:
> |
> | Subject: [Asterisk-Users] chan_sip.c problems problems from
> cvs 1.134 |
>
> >---
> |
>
>
>
>
> All,
>
> I've found problems in my setup with the latest couple of revisions
> (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9
> asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's,
> everything
> is in the same VLAN and only running SIP.
>
> Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300
>
> But inbound calls fail, I see the initial INVITE from the AS5300 which
> is
> received by asterisk but not responded to and then the AS5300 sends
> another
> few INVITE's which are received but ignored assumable as they were
> duplicates for the first.
>
> Unfortunately since I've been trying the different cvs revisions of
> chan_sip.c I've got susbequent problems with the server crashing after
> the
> first INVITE from the AS5300 using anything greater than cvs 1.134
>
> I suspect this is something to do with the per-user limits added in cvs
> 1.135 but I am curious to see if anyone has any problems with the latest
> cvs elease of asterisk with SIP ?
>
> Adam
>
> Sip read:
> INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0
> Via: SIP/2.0/UDP 213.160.252.50:53893
> From: "611012210"
> To:
> Date: Wed, 30 Jul 2003 09:26:11 GMT
> Call-ID: [EMAIL PROTECTED]
> Cisco-Guid: 1667049428-3407675953-0-149543808
> User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
> CSeq: 101 INVITE
> Max-Forwards: 6
> Timestamp: 1059557171
> Contact:
> Expires: 180
> Content-Type: application/sdp
> Content-Length: 149
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50
> s=SIP Call
> c=IN IP4 213.160.252.50
> t=0 0
> m=audio 20032 RTP/AVP 8 0 65535 18
>
> 15 headers, 6 lines
> Using latest request as basis request
> Sending to 213.160.252.50 : 53893 (non-NAT)
> Found audio format 8
> Found audio format 0
> Found audio format 65535
> Found audio format 18
> Capabilities: us - 524302, them - 268/0, combined - 12
> Non-codec capabilities: us - 1, them - 0, combined - 0
> AM00CM01*CLI>
> Disconnected from Asterisk server
>
>
> * DISCLAIMER *
>
> This message and any attachment are confidential and may be privileged
> or
> otherwise protected from disclosure and may include propriet
RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Brenton, Yves, ... I've located the cause of the problem in chan_sip.c but am still trying to find the exact cause being completely new to the asterisk code. It seems that there was an added function in 1.135 called 'find_user' that is supposed to lookup the users incoming call limit but the routine is unable to find a matching user for my AS5300 which I suspect is because it does not REGISTER with the server prior to attempting to send calls. I'm going to continue debugging a little later and see if I can narrow it down more ... Adam -Original Message- From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: 30/07/03 14:09 Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 Hi, I am using the latest cvs release of asterisk, and the behaviour is in fact the same, outbound calls work fine, but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked" by asterisk, and never reach the phone. The setup is the same : 7960 <--> asterisk <--> C2651<-> PSTN Yves |-+-> | | "Low, Adam" | | | <[EMAIL PROTECTED]>| | | Sent by: | | | [EMAIL PROTECTED]| | | .digium.com | | | | | | | | | 30/07/2003 11:37 | | | Please respond to | | | asterisk-users| | | | |-+-> >--- | | | | To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> | | cc: | | Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 | >--- | All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: "611012210" To: Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary informati
Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Hi, I am using the latest cvs release of asterisk, and the behaviour is in fact the same, outbound calls work fine, but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked" by asterisk, and never reach the phone. The setup is the same : 7960 <--> asterisk <--> C2651<-> PSTN Yves |-+-> | | "Low, Adam" | | | <[EMAIL PROTECTED]>| | | Sent by: | | | [EMAIL PROTECTED]| | | .digium.com | | | | | | | | | 30/07/2003 11:37 | | | Please respond to | | | asterisk-users| | | | |-+-> >---| | | | To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> | | cc: | | Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 | >---| All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: "611012210" To: Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI> Disconnected from Asterisk server * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Voip Gateway Config
Hello, Can anyone spare a configuration file for using asterisk as a pure Voip-PSTN gateway running H323 and Sip only ? Many thanks, Abdul Hakeem ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Dan, The time to call could be stored into database with party A and party B phone number. Asterisk or perhaps a script (mentions by Andy Powel in another reply) just keep checking the database and make calls if time is < current time and the call has not been processed yet. In this manner, the caller can even schedule a call for tomorrow mornnig, all he do is just insert a record in database and wait :). Foong - Original Message - From: "Dan" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 7:15 PM Subject: Re: [Asterisk-Users] Call Transfer > Foong, > > > Actually, we have a client that is too lazy to do all the dialing, he want > a > > system that will call him and also the person he wanted to call, just like > > some receptionists do theese days. The different is that asterisk is > taking > > over the receptionist's job > ... then... who decide when the call must be initiated and how? > > Dan > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Thanks Andy Will try that Thanks again. Foong - Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 6:56 PM Subject: Re: [Asterisk-Users] Call Transfer > Foong > > Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below > > Channel: SIP/[EMAIL PROTECTED] > MaxRetries: 2 > RetryTime: 60 > WaitTime: 30 > Context: mysipcontext2 > Extension: 2000 > Priority: 1 > > This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. > > All you need is a script to lookup in the database and generate the script file for you and it's done. > > HTH > > Andy > > > *** REPLY SEPARATOR *** > > On 30/07/2003 at 16:30 Chee Foong wrote: > > >Hello Dan, > > > >Thanks for you reply. > > > >Base on you recomendation using the 'T' argument. I manage to do call > >transfer an it works really well. > > > >My problem comes when my boss comes out with a superb idea where the > >transfering process is automated without involving a human :( > > > >Say asterisk get 2 numbers (from database, text file, etc), one belongs > >party A and the other belongs to party B. Asterisk will calls both parties > >and do the tranfer automatically. In another words, asterisk is resposible > >to 'press' the '#' to do the transfer. I don't this can be achieve in the > >extension.conf not matter how you structure you dial plan. > > > >Perhaps, the only way is to write a apps and plug it into asterisk like all > >the asterisk modules such as Meetme. > > > >Any ideas? > > > > > >Foong > > > >- Original Message - > >From: "Dan" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Wednesday, July 30, 2003 3:42 PM > >Subject: Re: [Asterisk-Users] Call Transfer > > > > > >> Hi, > >> > >> It works if you put the 'T' switch in the dial line. > >> > >> You can then transfer the call from the caller. > >> I have tested it in the folllowing configuration and it works: > >> Call from a Cisco 7960 to an ATA 186. > >> Select 'Transfer" on 7960 > >> Call another extension (X-Lite) > >> Select again transfer on 7960. > >> The call remain between ATA and X-Lite. > >> > >> This is what you need? > >> > >> BR, > >> Dan > >> > >> - Original Message - > >> From: "Chee Foong" <[EMAIL PROTECTED]> > >> To: <[EMAIL PROTECTED]> > >> Sent: Wednesday, July 30, 2003 7:08 AM > >> Subject: [Asterisk-Users] Call Transfer > >> > >> > >> Hello all, > >> > >> I am in a situation where I need to use asterisk to call someone say > >Party > >> A. After the call to Party A got through, asterisk will put Party A on > >hold, > >> then asterisk will call Party B. If call to Party B got through, asterisk > >> will transfer Party A to Party B. > >> > >> I wonder if this features is implemented into asterisk. I have found a > >post > >> in asterisk mailing list: > >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html > >> > >> but that doesn't help much. > >> > >> If this features is not implemented, can anyone give me some point on how > >to > >> implement this in asterisk? Do I need to write an app like the Dial apps > >for > >> asterisk to load at start up? > >> > >> > >> thanks > >> > >> Foong > >> > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
There is no need to create a Meeting Room... just to initiate a conference in three... - Original Message - From: "Chee Foong" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 1:02 PM Subject: Re: [Asterisk-Users] Call Transfer > Hello > > But If i do that I have to create lots of conference room if I have lots of > caller. > > Foong > > - Original Message - > From: "Sip Rtp" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, July 30, 2003 5:44 PM > Subject: Re: [Asterisk-Users] Call Transfer > > > > Yes, I second to that idea. > > I think thats only available option to put them in a > > local conference. > > Rgds > > Manoj K Gupta > > > > - Original Message - > > From: "Dan" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Wednesday, July 30, 2003 2:04 PM > > Subject: Re: [Asterisk-Users] Call Transfer > > > > > > > Hi Foong, > > > > > > But then... who and when will trigger the transfer > > between the two remote > > > extensions? > > > > > > I think to something like that. > > > One of the extension calls a special number, > > entering a password (or check > > > after the Caller ID). > > > Asterisk close the call, wait for answer > > > Call the second extension, wait for answer > > > Then, in some way (eventually through a conference > > mode using local > > CONSOLE > > > as master) bridge the two calls. > > > What do you think about that? > > > > > > Dan > > > > > > > > > - Original Message - > > > From: "Chee Foong" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > Sent: Wednesday, July 30, 2003 11:30 AM > > > Subject: Re: [Asterisk-Users] Call Transfer > > > > > > > > > > Hello Dan, > > > > > > > > Thanks for you reply. > > > > > > > > Base on you recomendation using the 'T' argument. > > I manage to do call > > > > transfer an it works really well. > > > > > > > > My problem comes when my boss comes out with a > > superb idea where the > > > > transfering process is automated without involving > > a human :( > > > > > > > > Say asterisk get 2 numbers (from database, text > > file, etc), one belongs > > > > party A and the other belongs to party B. Asterisk > > will calls both > > parties > > > > and do the tranfer automatically. In another > > words, asterisk is > > resposible > > > > to 'press' the '#' to do the transfer. I don't > > this can be achieve in > > the > > > > extension.conf not matter how you structure you > > dial plan. > > > > > > > > Perhaps, the only way is to write a apps and plug > > it into asterisk like > > > all > > > > the asterisk modules such as Meetme. > > > > > > > > Any ideas? > > > > > > > > > > > > Foong > > > > > > > > - Original Message - > > > > From: "Dan" <[EMAIL PROTECTED]> > > > > To: <[EMAIL PROTECTED]> > > > > Sent: Wednesday, July 30, 2003 3:42 PM > > > > Subject: Re: [Asterisk-Users] Call Transfer > > > > > > > > > > > > > Hi, > > > > > > > > > > It works if you put the 'T' switch in the dial > > line. > > > > > > > > > > You can then transfer the call from the caller. > > > > > I have tested it in the folllowing configuration > > and it works: > > > > > Call from a Cisco 7960 to an ATA 186. > > > > > Select 'Transfer" on 7960 > > > > > Call another extension (X-Lite) > > > > > Select again transfer on 7960. > > > > > The call remain between ATA and X-Lite. > > > > > > > > > > This is what you need? > > > > > > > > > > BR, > > > > > Dan > > > > > > > > > > - Original Message - > > > > > From: "Chee Foong" <[EMAIL PROTECTED]> > > > > > To: <[EMAIL PROTECTED]> > > > > > Sent: Wednesday, July 30, 2003 7:08 AM > > > > > Subject: [Asterisk-Users] Call Transfer > > > > > > > > > > > > > > > Hello all, > > > > > > > > > > I am in a situation where I need to use asterisk > > to call someone say > > > Party > > > > > A. After the call to Party A got through, > > asterisk will put Party A on > > > > hold, > > > > > then asterisk will call Party B. If call to > > Party B got through, > > > asterisk > > > > > will transfer Party A to Party B. > > > > > > > > > > I wonder if this features is implemented into > > asterisk. I have found a > > > > post > > > > > in asterisk mailing list: > > > > > > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html > > > > > > > > > > but that doesn't help much. > > > > > > > > > > If this features is not implemented, can anyone > > give me some point on > > > how > > > > to > > > > > implement this in asterisk? Do I need to write > > an app like the Dial > > apps > > > > for > > > > > asterisk to load at start up? > > > > > > > > > > > > > > > thanks > > > > > > > > > > Foong > > > > > > > > > > > > > > > ___ > > > > > Asterisk-Users mailing list > > > > > [EMAIL PROTECTED] > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PRO
Re: [Asterisk-Users] Call Transfer
Foong, > Actually, we have a client that is too lazy to do all the dialing, he want a > system that will call him and also the person he wanted to call, just like > some receptionists do theese days. The different is that asterisk is taking > over the receptionist's job ... then... who decide when the call must be initiated and how? Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer
Foong Take a look at the sample.call file, modifying the settings in there and copying the file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example config is below Channel: SIP/[EMAIL PROTECTED] MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: mysipcontext2 Extension: 2000 Priority: 1 This will make asterisk dial exten 1000 in the context mysipcontext when it's answered it will then call exten 2000 in mysipcontext2.. All you need is a script to lookup in the database and generate the script file for you and it's done. HTH Andy *** REPLY SEPARATOR *** On 30/07/2003 at 16:30 Chee Foong wrote: >Hello Dan, > >Thanks for you reply. > >Base on you recomendation using the 'T' argument. I manage to do call >transfer an it works really well. > >My problem comes when my boss comes out with a superb idea where the >transfering process is automated without involving a human :( > >Say asterisk get 2 numbers (from database, text file, etc), one belongs >party A and the other belongs to party B. Asterisk will calls both parties >and do the tranfer automatically. In another words, asterisk is resposible >to 'press' the '#' to do the transfer. I don't this can be achieve in the >extension.conf not matter how you structure you dial plan. > >Perhaps, the only way is to write a apps and plug it into asterisk like all >the asterisk modules such as Meetme. > >Any ideas? > > >Foong > >- Original Message - >From: "Dan" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Wednesday, July 30, 2003 3:42 PM >Subject: Re: [Asterisk-Users] Call Transfer > > >> Hi, >> >> It works if you put the 'T' switch in the dial line. >> >> You can then transfer the call from the caller. >> I have tested it in the folllowing configuration and it works: >> Call from a Cisco 7960 to an ATA 186. >> Select 'Transfer" on 7960 >> Call another extension (X-Lite) >> Select again transfer on 7960. >> The call remain between ATA and X-Lite. >> >> This is what you need? >> >> BR, >> Dan >> >> - Original Message - >> From: "Chee Foong" <[EMAIL PROTECTED]> >> To: <[EMAIL PROTECTED]> >> Sent: Wednesday, July 30, 2003 7:08 AM >> Subject: [Asterisk-Users] Call Transfer >> >> >> Hello all, >> >> I am in a situation where I need to use asterisk to call someone say >Party >> A. After the call to Party A got through, asterisk will put Party A on >hold, >> then asterisk will call Party B. If call to Party B got through, asterisk >> will transfer Party A to Party B. >> >> I wonder if this features is implemented into asterisk. I have found a >post >> in asterisk mailing list: >> http://lists.digium.com/pipermail/asterisk-users/2003-June/013253.html >> >> but that doesn't help much. >> >> If this features is not implemented, can anyone give me some point on how >to >> implement this in asterisk? Do I need to write an app like the Dial apps >for >> asterisk to load at start up? >> >> >> thanks >> >> Foong >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Random Hangup Problems
Hello, This morning I just started to have this problem calling from a SIP phone to a regular phone, using one of the 4 BRI cards (passive) I've in my * box. It calls regularly, but somewhere after 8-10 secs, it random hangups, or it hangups immediately after a hold, and so. I've looked into /var/log/asterisk/messages, and this is the output corresponding to the hangup: # Jul 30 12:26:22 WARNING[11276]: File chan_sip.c, Line 417 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Jul 30 12:26:27 WARNING[11276]: File chan_sip.c, Line 2156 (__transmit_response): Unable to determine sequence number from '' # Any suggestion? I'm quite disoriented in this moment, i initially tought that was a problem due to the "t" command for the Dial app, (somewhere it played "pbx-invalid" and hung up), but that part is now solved... still remaining the random hangups :( -- Stefano Finetti ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Microsoft SQL
> As suggested by another poster: MS-SQL is mostly based on Sybase, so any > Sybase driver (there is one for PHP for instance) can probably be used, > from AGI or otherwise... I wrote a check_mssql perl script for nagios (dot org). Take a look at this for reference - it really isn't hard :) http://sourceforge.net/tracker/index.php?func=detail&aid=738128&group_id=29880&atid=541465 -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 2254 5070 (work) +47 9801 3356 (mobile) Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SetCIDName
chan_h323 passes Caller*id, if there is one. Then you can specify a type=h323 for a specific H.323ID, if you wish. Jeremy McNamara Tais M. Hansen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 30 July 2003 11:58, Jeremy McNamara wrote: Because H.323 doesn't have a specific 'feature' of caller*id. It's pretty useless always seeing "root" in the phone's display no matter who's calling. Anyway to get around it? - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/J5nT2TEAILET3McRAnprAJ42dxXWweL9RWGBKd0DJDwuV6RBHgCgiO0A wl3/X2fP4/lq+aU1JaK61xI= =dYqc -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
