Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
No its stable just not as featureful as head. The tag is the same and you can still check it out. bkw - Original Message - From: Chris Foster [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 12:32 AM Subject: Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch? On Mon, 5 Jul 2004 22:02:37 -0700 (PDT), every buddy [EMAIL PROTECTED] wrote: A while ago on the download page on www.asterisk.org, there was a stable branch for the asterisk source tree. It seems to have disappeared now, at least the instructions on that web page are gone. What's the story on this? Can we have it back please? thanks stable's gone because it wasn't too stable. The lastest CVS source is alot more full featured and stable then the old stable branch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8# into invalid extensions
ok. Thank U for a hint. I have find out, the problem was with my ATA-186. That box just use '#' not as sending key. Does anyone know how to force ATA-186 to use '#' as sending key. Have tried *8 from softphone and that works fine. On Mon, 2004-07-05 at 21:19, Brancaleoni Matteo wrote: Hi Il lun, 2004-07-05 alle 20:12, Brian K. West ha scritto: *8# works on sip that uses the # as the send key. sure, but since he gets -- Sent into invalid extension '*8#' in context 'from-sip-post'... means that he's sending *8# ... matteo -- Best regards Vlad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any experience with Citel Link 3300 and Asterisk
Hello I am currenly trying to setup Citel Link with Asterisk. So far i don't have any luck in able to assign IP address to Citel Link channel but still thought not a bad idea and get some one's view about Citel channel bank. Please let me know your experience while setting up Citel and Asterisk. I will appreciate if you tell me the way to talk to Citel Link using Hyper Terminal. Thanks Deepak
Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
On Tue, 6 Jul 2004 00:32:20 -0500, Chris Foster [EMAIL PROTECTED] wrote: stable's gone because it wasn't too stable. The lastest CVS source is alot more full featured and stable then the old stable branch. I've found it the opposite. I've tried CVS Head a few times because I wanted some of the latest features, but every time I have gone back to the trusty Stable CVS build we've been using on two machines. Not once has CVS Stable gone down or had any problems. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Calling an outside phone number as part of a hunt
On Mon, 05 Jul 2004 22:38:22 -0500, Daniel Jimenez [EMAIL PROTECTED] wrote: Hall, Eric M. wrote: I'm trying to see if this is even possible. AFAIK Asterisk has no way of knowing if you do not answer. To Asterisk, the call is complete and answered when it starts ringing. A PSTN/POTS call is always going to be the final destination. With Analogue interfaces (X100P, etc) - yes, a call is marked ANSWERED as soon as it starts ringing. It's a different story with ISDN/digital interfaces. On several occasions I've set my desk phone to ring with my cell phone, etc. - the first one to answer gets the call. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk, fwd, and grandstream?
can this be accomplished? Yes. You should start reading documentation before asking. A good starting place is http://www.voip-info.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
On Tue, Jul 06, 2004 at 12:32:20AM -0500, Chris Foster wrote: Date: Tue, 6 Jul 2004 00:32:20 -0500 From: Chris Foster [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch? On Mon, 5 Jul 2004 22:02:37 -0700 (PDT), every buddy [EMAIL PROTECTED] wrote: A while ago on the download page on www.asterisk.org, there was a stable branch for the asterisk source tree. It seems to have disappeared now, at least the instructions on that web page are gone. What's the story on this? Can we have it back please? thanks stable's gone because it wasn't too stable. The lastest CVS source is alot more full featured and stable then the old stable branch. Disagree. At least I had Dlink DPH-100M (mgcp phone) working fine with stable. Whith cvs head it is working strange. Doesn't provide tone on handset pickup, strips 1st dialed digit, etc. -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
Brian K. West [EMAIL PROTECTED] wrote: No its stable just not as featureful as head. The What is important to me is the fact that I have the same known release on multiple installations, whether it's more stable or not. I don't want to have a different release on every machine I look after. tag is the same and you can still check it out. That's good news. Unfortunately I don't seem to have any record anymore, Always looked it up on the web site. Would you care to post here what the command was again to get the stable branch from CVS? thanks. __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2x analog interface (1 ISDN and 1 door phone) recomendation for Europe ?
Hi, I'd like to use Asterisk with ISDN interface and normal analog interface to door phone (or any other low cost connection type to door phone). What would be your recomendations for needed HW in Europe? Is it possible to have this in one PCI card? Are there any lower cost voip door phones? Thanks in advance, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to connect to cellular phone beside analog interface card?
Hi, I have SE P800 cellular phone and I'm curious whether I could connect to it over internet and not over analog interface and GSM network. Are there any other cellulars that can do this ? Which ones ? Thanks in advance, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
On Tue, 6 Jul 2004 00:18:15 -0700 (PDT), every buddy [EMAIL PROTECTED] wrote: That's good news. Unfortunately I don't seem to have any record anymore, Always looked it up on the web site. Would you care to post here what the command was again to get the stable branch from CVS? thanks. It was: cvs checkout -r v1-0_stable asterisk So, assuming the tag is the same, that should work. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2x analog interface (1 ISDN and 1 door phone) recomendation for Europe ?
http://www.voip-info.org/wiki-Asterisk+phone+door might be of some use. -Shaun On Tue, 6 Jul 2004 09:27:07 +0200, Robert Rozman [EMAIL PROTECTED] wrote: Hi, I'd like to use Asterisk with ISDN interface and normal analog interface to door phone (or any other low cost connection type to door phone). What would be your recomendations for needed HW in Europe? Is it possible to have this in one PCI card? Are there any lower cost voip door phones? Thanks in advance, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again on the ZyXEL Prestige 2000W
It is. I did a cross upgrade with Pulver's firmware. I could not notice any improvements, though... I still had that annoying hangup problem. lenz wrote: I have heard that the 2000W is the same exact harware as the PulverInnovations WiSip phone - http://www.pulverinnovations.com/ - so the drivers might be the same, but I have not tried this. dominique kull taridium.communications the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn to sip callerID pass
hi, I have a problem with passing caller id information from telco (isdn) to sip client (grandstream). i see callerid in asterisk verbose console but on grandstream (sip) phone is just internal (own-gs) 101 number. Isdn line is connected with hfc card and p2p , asterisk is latest CVS in extensions.conf i have: exten = 2442242,1,Dial,SIP/101,r|T and in console is this: -- Executing Dial(Zap/1-1, SIP/101) in new stack -- Called 101 -- Accepting call from '4482333' to '2442242' on channel 1, span 1 -- SIP/101-b1e2 is ringing -- Channel 1, span 1 got hangup and number on GS sip phone must be 4482333 but is only 101 ! maybe i missed something? thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk, fwd, and grandstream?
I have all ready been there the only refference I saw was the tips and tricks for asterisk and grandstream is there some info I am missing? thanks hank - Original Message - From: Holger Schurig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 12:06 AM Subject: Re: [Asterisk-Users] asterisk, fwd, and grandstream? can this be accomplished? Yes. You should start reading documentation before asking. A good starting place is http://www.voip-info.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is srv lookup being done when REGISTERing?
it looks (to me) like asterisk is not doing an SRV lookup when REGISTERing with another sip proxy. is that correct? what i am trying to achieve is to register [EMAIL PROTECTED] with a proxy using register = jasko:secret:[EMAIL PROTECTED] my problem is that asterisk is doing a simple A RR lookup for the domain telia.net which is pointing to a host that is NOT the proxy for that domain (resulting in the REGISTER message ending up with the wrong host). if an SRV lookup had been done instead, the REGISTER message would have be sent to the right host. [i cannot change telia.net in the above line as that messes up the authentication instead. and i do not have control over the dns for the domain in question]. is there any other way to force the REGISTER message through a certain proxy/host? am i missing something? jasko ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
Shaun Ewing [EMAIL PROTECTED] wrote: It was: cvs checkout -r v1-0_stable asterisk thanks a lot. __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is srv lookup being done when REGISTERing?
Jasminko Mulahusic wrote: it looks (to me) like asterisk is not doing an SRV lookup when REGISTERing with another sip proxy. is that correct? what i am trying to achieve is to register [EMAIL PROTECTED] with a proxy using register = jasko:secret:[EMAIL PROTECTED] my problem is that asterisk is doing a simple A RR lookup for the domain telia.net which is pointing to a host that is NOT the proxy for that domain (resulting in the REGISTER message ending up with the wrong host). if an SRV lookup had been done instead, the REGISTER message would have be sent to the right host. [i cannot change telia.net in the above line as that messes up the authentication instead. and i do not have control over the dns for the domain in question]. is there any other way to force the REGISTER message through a certain proxy/host? am i missing something? jasko This issued has been discussed few weeks ago into great depth. Look into May/June 04 archive! Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I disable '#' to transfer a call?
I don't see anything on the Wiki or in the documentation about disabling this feature. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk, fwd, and grandstream?
I have all ready been there the only refference I saw was the tips and tricks for asterisk and grandstream Do it the roman way: Divide and conquer. Divide your problems into 3 little problems: a) connect a Grandstream to Asterisk b) connect Asterisk to Grandstream c) dialplan magic to connect them to each other And then conquer them problem by problem. a) is describe in the sip.conf.sample that comes with asterisk as well as in the WIKI. If you have the source code of asterisk, look into the configs/ directory. Hey, you can even search for grandstream. b) Is in the Wiki explained. You can search for freeworlddialup and get http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD as a result. c) is explained in this document as well at the end. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I disable '#' to transfer a call?
Easy, just don't include t or T in the dial string options. -Shaun On Tue, 06 Jul 2004 01:38:23 -0700, Dameon D. Welch-Abernathy [EMAIL PROTECTED] wrote: I don't see anything on the Wiki or in the documentation about disabling this feature. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
We're using the Quad-BRI card from Junghanns.NET with corresponding drivers (bristuff 0.0.2). The driver tries to patch asterisk libpri, which fails for current version. Anyone got an idea what'S the latest version of asterisk / libtri usable with the Quad-BRI Card? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I disable '#' to transfer a call?
I don't see anything on the Wiki or in the documentation about disabling this feature. What about the product documentation? Certainly your phone has some means of configuration, e.g. by config files, built-in menus or a web-browser. Use that and the documentation for it. Maybe I'm wrong, but I see the wiki more as a documentation for Asterisk and for Hints on how to use Asterisk with various products. It is *NOT* a substitute for the product documentation. Trying to keep track all possible products and their various firmware releases would be really tedious :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 Call Transfers
We're using a couple of h323 IP Phones (innovaphone ip200) w/ asterisk. Basic call setup works, but we can't get call transfers to work: On pressing the transfer button on the phone (getting a new dialtone) the 2nd endpoint is disconnected. Any idea if we can get this to work? Same reaction using the innovaphone ip400 gatekeeper and using gnugk. Asterisk version is 0.7.2 release. Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZyXEL P2000W - working conf example
(I posted this note on http://www.voip-info.org/wiki-ZyXEL+P2000W+configuration too) I tried to put together comments that were asked for on the P2000W. These configs seem to work fine for a ZyXEL P2000W, thanks to Giles Scott for getting me started with it. DTMF keys work fine and are read correctly by Asterisk. It's important that you upgrade your phone to a modern version of the firmware - it didn't work much with the WJ.00.07 that was preinstalled on my terminal, while it works with WJ.00.0c See: http://www2.studerus.ch/support.cfm?action=newslang=d http://www.nikotel.net/firmware/zyxel/p2000w/P2000W_WJ000E_Standard.zip In this example, my phone has number 898 and 10.10.3.5 is Asterisk. Cheers l. sip.conf [898] type=friend username=phone secret=---mypassword--- host=dynamic canreinvite=no context=sip disallow=all allow=alaw dtmfmode=rfc2833 On the telephone web interface: Zyxel config SIP/outbound Proxy config Proxy IP:10.10.3.5 Proxy port = 5060 SIP Config SIP URI: sip: 898 @ 10.10.3.5 : 5060 Expire time: 300 Registrar username: 895 Registrar password: ---mypassword--- DSP setting Default Voice codec G.729 8k DTMF relay: outband -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
Hi Martin, The bristuff distribution comes with a install.sh script (./install.sh) which downloads, compiles the required software on your system. If you want to do it manually, look into download.sh to see the exact cvs checkout options which downloads the required asterisk and libpri versions. Regards, Michael Martin Bene wrote: We're using the Quad-BRI card from Junghanns.NET with corresponding drivers (bristuff 0.0.2). The driver tries to patch asterisk libpri, which fails for current version. Anyone got an idea what'S the latest version of asterisk / libtri usable with the Quad-BRI Card? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 Call Transfers
Martin Bene wrote: Asterisk version is 0.7.2 release. How about running a current (cvs -head) version of Asterisk? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
The bristuff distribution comes with a install.sh script (./install.sh) which downloads, compiles the required software on your system. If you want to do it manually, look into download.sh to see the exact cvs checkout options which downloads the required asterisk and libpri versions. Yes, I know which libpri/asterisk versions bristuff downloads when using the included scripts (03/24/04). Problem is, I'd like to get the features / bugfixes from later versions. I'd especially like to try current oh323 drives, which require cvs head and don't compile against the versions usd by bristuff 0.2.2. Is it possible to combine older libtri with cvs-head asterisk or is that just asking for trouble? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: is srv lookup being done when REGISTERing?
This issued has been discussed few weeks ago into great depth. Look into May/June 04 archive! i have indeed looked into archives (wiki, googled using tabs, asked my barber) and what have been discussed were SRV records for outgoing calls. the same seems not to work for REGISTER. jasko ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] H323 Call Transfers
How about running a current (cvs -head) version of Asterisk? Would love to and of course tried to: no go because of Junghans Quad-BRI ISDN Card, no driver for cvs -head. Bye, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] H323 Call Transfers
Martin Bene wrote: How about running a current (cvs -head) version of Asterisk? Would love to and of course tried to: no go because of Junghans Quad-BRI ISDN Card, no driver for cvs -head. Then complain to Junghans. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: is srv lookup being done when REGISTERing?
Jasminko Mulahusic wrote: This issued has been discussed few weeks ago into great depth. Look into May/June 04 archive! i have indeed looked into archives (wiki, googled using tabs, asked my barber) and what have been discussed were SRV records for outgoing calls. the same seems not to work for REGISTER. Well, I have not followed that thread into great details but my understanding is that * does not support SRV records properly! If you find a concrete answer I would be interested to know about it! BTW... Jasko, gdje se ti nalazis/radis? Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
Then wait for the next version, which will support both branches. If you can't wait you can use the patch from someone who merged the bristuff patch with a more recent version of cvs head... This one: http://capi4linux.thepenguin.de/download/asterisk/bri-stuff-0.0.2a-pp.tar.gz Michael Martin Bene wrote: The bristuff distribution comes with a install.sh script (./install.sh) which downloads, compiles the required software on your system. If you want to do it manually, look into download.sh to see the exact cvs checkout options which downloads the required asterisk and libpri versions. Yes, I know which libpri/asterisk versions bristuff downloads when using the included scripts (03/24/04). Problem is, I'd like to get the features / bugfixes from later versions. I'd especially like to try current oh323 drives, which require cvs head and don't compile against the versions usd by bristuff 0.2.2. Is it possible to combine older libtri with cvs-head asterisk or is that just asking for trouble? Thanks, Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 channel
Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No need to say that H323-H323 is working, as well as SIP-SIP. Running CVS version from yesterday. Used codecs are G711U A, G723.1 and G729. If I just use G711 it's the same. SIP EP has to call first when * is started to make it work. Any hint? Also, H323 is still broken and working without FastStart. Is there a workaround existing? Regards -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Protocol Error (6) using Zaphfc
Just wanted to say, that the problem was codec-related (on sipphone
connected to *)
Changed codec-settings and zaphfc is now running cool with cvs head.
- Still I don't know why I got the Protocol error (6) - but who cares?! :-)
NRB
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 1:56 PM
Subject: Re: [Asterisk-Users] Protocol Error (6) using Zaphfc
Hei,
please never try to dial out on a particular b channel, you have to dial
out on a zaptel group which includes both b channels of the BRI line.
In a p2mp setup YOU cannot know which b channel will be chosen!
exten = _X.,1,Dial(ZAP/g1/${EXTEN})
will do(note the 'g')
best regards
Klaus
Am Mo, 2004-06-28 um 12.45 schrieb nrb:
Hi!
Has anybody seen anything like this using zaphfc?
On outgoing calls (via isdn) , the line gets hung-up as soon as the
called
party answers.
As seen below i get some protocol error (6) - but i'm not sure if this
is
related to the hang-up which apparently comes a little earlier?!
Incomming calls on the isdn (zaphfc) interface is working just fine
(P.S. what about the D-channel going up down all the time - is that
normal? )
Kind Regards
NRB
Setup
Bri-stuff - 0.0.20
Asterisk CVS-HEAD-06/23/04-15:45:48 built by
[EMAIL PROTECTED] on a
i686 running Linux
Zapata.conf:
[channels]
switchtype = euroisdn
; p2mp TE mode
signalling = bri_cpe_ptmp
; p2p TE mode
;signalling = bri_cpe
; p2mp NT mode
;signalling = bri_net_ptmp
; p2p NT mode
;signalling = bri_net
pridialplan=local
prilocaldialplan=local
echocancel=yes
immediate=yes
group = 1
context=demo
channel = 1-2
Zaptel.conf:
loadzone=nl
defaultzone=nl
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
Example where a sip client (2203) is calling 7024
From Asterisk:
== D-Channel on span 1 down
== D-Channel on span 1 up
-- Executing Dial(SIP/2203-5779, Zap/1/7024) in new stack
-- Making new call for cr 135
Protocol Discriminator: Q.931 (8) len=32
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: SETUP (5)
Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer
capability:
Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number passed network
screening
(1) '2203' ]
Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '7024' ]
Sending Complete (len= 0)
-- Called 1/7024
Protocol Discriminator: Q.931 (8) len=7
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CALL PROCEEDING (2)
Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0,
Exclusive
Dchan: 0
ChanSel: B1 channel
]
-- Processing IE 24 (Channel Identification)
Protocol Discriminator: Q.931 (8) len=12
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: ALERTING (1)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Inband information or appropriate
pattern now
available. (8) ]
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Network beyond the interworking point (10)
Ext: 1 Progress Description: Unknown (1) ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 30 (Progress Indicator)
-- Zap/1-1 is ringing
Protocol Discriminator: Q.931 (8) len=15
Call Ref: len= 1 (reference 135/0x87) (Terminator)
Message type: CONNECT (7)
Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard
(0) 0: 0
Location: Public network serving the remote user (4)
Ext: 1 Progress Description: Unknown (4) ]
Time Date (len= 5) [ 04-06-28 11:58 ]
-- Processing IE 30 (Progress Indicator)
-- Processing IE 41 (Date/Time)
Protocol Discriminator: Q.931 (8) len=4
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: CONNECT ACKNOWLEDGE (15)
-- Zap/1-1 answered SIP/2203-5779
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate
Connect
Request
Protocol Discriminator: Q.931 (8) len=8
Call Ref: len= 1 (reference 7/0x7) (Originator)
Message type: DISCONNECT (69)
Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location:
Private network serving the local user (1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
-- Hungup 'Zap/1-1'
== Spawn extension (intern, 7024, 1) exited non-zero on
'SIP/2203-5779'
Protocol Discriminator: Q.931 (8) len=4
Call Ref:
AW: AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
If you can't wait you can use the patch from someone who merged the bristuff patch with a more recent version of cvs head... This one: http://capi4linux.thepenguin.de/download/asterisk/bri-stuff-0. 0.2a-pp.tar.gz Thanks for that pointer, I'll give it a try. Bye, martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 channel
OH323 seems to work... Might be an alternative Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of administrator tootai Sent: Tuesday, July 06, 2004 3:23 AM To: Asterisk-Users Subject: [Asterisk-Users] H323 channel Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No need to say that H323-H323 is working, as well as SIP-SIP. Running CVS version from yesterday. Used codecs are G711U A, G723.1 and G729. If I just use G711 it's the same. SIP EP has to call first when * is started to make it work. Any hint? Also, H323 is still broken and working without FastStart. Is there a workaround existing? Regards -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No RED/GREEN alerts on TDM400P?
I replaced my X100P cards with two TDM04B fully populated (8 FXO modules). They are working fine, I can make and receive calls, but I noticed all modules are always in GREEN state, even if I disconnect the line. Both zttools and a cat /proc/zaptel/device shows no RED alarm. Is there a workaround for this? There is no workaround. I opened a bug on this subject (well, the bug was oriented around * not knowing when a pstn line was down/disconnected and would continue to send calls out the zap port), and Mark closed it with comments something like... that's by design, will never be fixed. Not sure what the logic is behind that, but it certainly does _not_ emulate any typical pbx or switch that I've ever seen in 20+ years of telephony. The fxo module has all of the hardware logic to sense this (as well as many other things), but for whatever reason, no one wants to deal with implementing the software logic to support it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
Yes, I know which libpri/asterisk versions bristuff downloads when using the included scripts (03/24/04). Problem is, I'd like to get the features / bugfixes from later versions. In the wiki (and the mailing list archive) there's a document how I got Asterisk CVS working with bri-stuff. The trick was to use the download.sh script as normal. Then I copied the chan_zap file from the older asterisk version into the CVS version, fixed the thread stuff and voila, it worked. At least with my HFC card. Drawback: chan_zap got some fixes in the last time which you won't have using this approach. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calling an outside phone number as part of a hunt
Thank you! That's what I was thinking but being new I wanted to ask . Thanks again -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Jimenez Sent: Monday, July 05, 2004 11:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Calling an outside phone number as part of a hunt Hall, Eric M. wrote: I'm trying to see if this is even possible. AFAIK Asterisk has no way of knowing if you do not answer. To Asterisk, the call is complete and answered when it starts ringing. A PSTN/POTS call is always going to be the final destination. -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P PINS
On Tue, 2004-07-06 at 01:49, Wolfgang Pichler wrote: Am Di, den 06.07.2004 schrieb Steven Critchfield um 0:54: On Mon, 2004-07-05 at 06:51, Wolfgang Pichler wrote: hi all, Am Fr, den 02.07.2004 schrieb Steven Critchfield um 17:20: Chill down a bit. We here to help. sorry for that - but its really driving me crazy that i can't get it working (really tryied everything - there was already someone from our telco here - and a technican who has a digium card already running here in austria - no one seems to know where the failure is) Okay, the only thing I can think of now is to find out how long your connection is to the smart jack? One of our last installs had a 100+ foot length and it was really picky about the cable. We had one that was okay for ISDN but not for PRI over the same exact distance. my connection to the smart jack is about 5 foot length - shouldn't be a problem. Remember that you need a standard straight through cable to the PSTN. we are already using a straight through cable (also tryied a cross over cable) also if i had already tested the card with a loop cable - could it be possible that the card has a failure ? You will have to change signalling to something like a channelized T1 to use a loopback, I think. The PRI has complementary protocols for CPE and NET sides of the link. Not sure if a loopback would come up if it is configured for CPE. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing # on a crisco (was: Divert to arbitrary number)
Is it possible to have a speed dial on a cisco 7960 which dials the voice
mail number and then dials the extention and password so a user can
just push a single button to get their voicemail? This is a no brainer
on a regular analog phone where you can do something like 8500p100p1234
where p is a pause that most analog phones let you place in a speeddial,
but on a sip phone how is this done?
You do that on Asterisk in extensions.conf, not on the phone itself.
Either you simply do not set a pw in voicemail.conf, or you check the
caller ID of the phone and then decide if you allow or reject access to a
special extension that calls VoiceMailMain with the 's' option.
In the Cisco 7960 SIPmac-addr.cnf file, add a statement like:
messages_uri: 1234
and in your extensions.conf, add something like:
exten = 1234,1,Wait,1
exten = 1234,2,VoicemailMain(s${CALLERIDNUM})
When you press the messages button on the front of the 7960,
you'll go directly to voicemail bypassing the user input for the
password.
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[Asterisk-Users] fax detection and X100P
i have successfully updated my cvs pull of zaptel but for asterisk when i type "make clean"i have the folowing error: Makefile:73: *** missing separator. Arrêt ( Arrêt means stop) Lamine
[Asterisk-Users] Asterisk config on PostgreSQL
Hi all, Based on the website http://svn.asteriskdocs.org/ast_data, I am trying to migrate my config file to PostgreSQL,but I am having problem calling the other endpoint which I configured his account and extension on sql. I got an error code 484. I would like to ask what should be the correct entry in the extension in sql. regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax detection and X100P
if you 're not using linux you have to use gmake, not make Jean-Yves On 06/07/2004, at 9:22 PM, Mamadou Lamine KA wrote: i have successfully updated my cvs pull of zaptel but for asterisk when i type make cleani have the folowing error: Makefile:73: *** missing separator. Arrêt ( Arrêt means stop) Lamine
[Asterisk-Users] How to differentiate incoming calls with grandstream phone
Hello I've finally made the switch from our old PABX (NEC) to an Asterisk based server. I've configured zapatel to go into an incoming profile it gets into: extension = s,1,Dial(SIP/phone1SIP/phone2SIP/phone3... etc.. So when there's an incoming call, all phones rings then it goes into a specific voicemail etc... The problem is that it's impossible to tell weither it's an incoming call from outside (through a TMB03 card) or a called transfer from another SIP phone, the grandstream 101 phone only shows: TR1 (or something like that it's hard to tell with this display). Is there a way to configure either Asterisk or this phone to show a different display depending on the origin of the call? Thank you Jean-Yves ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] missing .gsm in VoiceMailMain(2)
Hi Folks I try the following within context: exten = foo,foo,VoiceMailMain After providung MailBoxNumber I get asked for PassWord. If now the input fails I see on CLI Playing 'vm-incorrect' followed by Playing 'vm-password' and I can hear both messages. Next try is: exten = foo,foo,VoiceMailMain(MBNumber) which jumps straight to the MB and asks my for PW If now the input fails I see on CLI Playing 'vm-incorrect' followed by Playing 'vm-password' but I can hear only the 'vm-password' message. So, the sample is there, CLI shows that it is played but I can't hear anything! Version is CVS-04/25/04-01:56:59 Any hints? Thanks -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cut off after 8 secs?? Help
Call comes in from remote SIP, authorised, does the following and dies Any idea why.. I have ports 5060 and 16384 to 16482 open Do I need any others? What am I missing Remote user is using X-lite for windows.. -- Executing Dial(SIP/2004-944c, SIP/2001|20) in new stack -- Called 2001 -- SIP/2001-4f3b is ringing -- SIP/2001-4f3b answered SIP/2004-944c -- Attempting native bridge of SIP/2004-944c and SIP/2001-4f3b Jul 5 19:04:17 WARNING[-224801872]: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 24922 (Response) It is dying because the audio stream (rtp packets) aren't getting through. Not sure why you picked rtp ports 16384-16482; each sip phone vendor picks there own set of port ranges, and the Xlite product use to use ports in the 8000 range (haven't checked lately). Read the stuff on the wiki relative to NAT parameters (for *) and you should be able to get it to work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] missing .gsm in VoiceMailMain(2)
On 06/07/2004, at 10:00 PM, Thomas Niesel wrote: I try the following within context: exten = foo,foo,VoiceMailMain After providung MailBoxNumber I get asked for PassWord. If now the input fails I see on CLI Playing 'vm-incorrect' followed by Playing 'vm-password' and I can hear both messages. Next try is: exten = foo,foo,VoiceMailMain(MBNumber) which jumps straight to the MB and asks my for PW Did you at least configure the mailbox and mailbox password in the mailbox.conf file? What did you define foo as? it has to be a number Jean-Yves ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to differentiate incoming calls with grandstream phone
Hi,
On 06/07/2004, at 9:52 PM, Jean-Yves Avenard wrote:
the grandstream 101 phone only shows: TR1 (or something like that it's
hard to tell with this display).
I had this problem for a while - the phone is actually displaying the
work asterisk in lower case - but it can't.
The GrandStream 101's display the number part of the caller id on their
display. So - you can get around this by setting the caller id before
you dial the extension.
Is there a way to configure either Asterisk or this phone to show a
different display depending on the origin of the call?
How you go about differentiating - that's a bit harder - but you could
set the caller id in your incoming context, and set it when you
transfer to a specific extension from the local context... or use a
goto statement based on the ${CHANNEL} variable... I haven't got this
working.
Also - try setting callerid = Name number in the sip.conf file for
each of the phones.
Andrew
_
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Real World Technology Solutions
Real People, Real SolUtions (tm)
ph: (02) 9945 2567 fax: (02) 9945 2566
mob: 0405 15 2568
http://www.rwts.com.au/
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[Asterisk-Users] Music on hold error since CVS update
Ever since i updated from CVS head i get this error when trying to play music on hold. res_musiconhold.c:314 moh0_exec: Unable to start music on hold (class '') on channel SIP/3001-f3b6 any ideas? Best Regards Stuart Baggs(Sales Manager) Web: www.t-hosting.bizEmail: [EMAIL PROTECTED]
[Asterisk-Users] * and Innovaphone
Hello,
I think I have the same problem as Martin Bene mentioned in
http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html
Since I found no further information about this I'd like to ask wether
you know what the reason for this problem is and how one can get around
this.
* is registered to the innovaphone gatekeeper.
Trunk connection is done with an AVM-B1 and chan_capi.
Regards
Torsten Krueger
Call signalling works fine, but in H323-Trace the following is shown:
2:21:03.498H245:8135df8 h323neg.cxx(835) H245
Received open channel: R-1, state=Released
2:21:03.509H245:8135df8 h323.cxx(4419) H323
Bandwidth request: -0.0kb/s, available: 32.0kb/s
2:21:03.510H245:8135df8 h323.cxx(4419) H323
Bandwidth request: -0.0kb/s, available: 32.0kb/s
2:21:03.511H245:8135df8 h323.cxx(4082) H245
Received early start OLC, aborting fast start
2:21:03.513H245:8135df8 h323.cxx(4179) H323
CreateLogicalChannel - forward channel
2:21:03.514H245:8135df8 h323caps.cxx(1824) H323
FindCapability: audioData
2:21:03.515H245:8135df8 h323caps.cxx(783) H323
Capability tx frames left at 30 as remote allows 60
2:21:03.526H245:8135df8 h323caps.cxx(1871) H323
Found capability: G.711-ALaw-64k{sw} 1
2:21:03.528H245:8135df8 h323caps.cxx(778) H323
Capability rx frames reduced from 240 to 60
2:21:03.530H245:8135df8 codecs.cxx(1062) Codec
G711 ALaw decoder created for at 64k, 480 samples
2:21:03.531H245:8135df8 channels.cxx(777) LogChan
Bandwidth requested/used = 64.0/0.0 kb/s
2:21:03.533H245:8135df8 h323.cxx(4419) H323
Bandwidth request: -0.0kb/s, available: 32.0kb/s
2:21:03.534H245:8135df8 h323.cxx(4419) H323
Bandwidth request: +64.0kb/s, available: 32.0kb/s
2:21:03.536H245:8135df8 h323.cxx(4425) H323
Available bandwidth exceeded
2:21:03.538H245:8135df8 h323.cxx(4225) H323
CreateLogicalChannel - insufficient bandwidth
2:21:03.539H245:8135df8 h323.cxx(4419) H323
Bandwidth request: -0.0kb/s, available: 32.0kb/s
2:21:03.543H245:8135df8 h323pdu.cxx(494) H245
Sending PDU:
response openLogicalChannelReject {
forwardLogicalChannelNumber = 1
cause = insufficientBandwidth null
Additional information:
Asterisk version used is: Asterisk CVS-HEAD-06/21/04-20:44:41 built by
[EMAIL PROTECTED] on a i686 running Linux
Innovaphone Versions used are
V4.00 sr5 IP400[03-4292] for the IP400 gateway that also runs the
gatekeeper and V4.00 sr5 IP200[03-4292] for the IP-Phone.
--
Media Online Internet Services Marketing GmbH
Torsten Krueger [EMAIL PROTECTED]
fon: 49-231-5575100fax: 49-231-55751098
Kurze Str. 10 D-44137 Dortmund
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Re: [Asterisk-Users] missing .gsm in VoiceMailMain(2)
Hallo Jean-Yves Avenard On Tue, 6 Jul 2004 22:08:39 +1000 you wrote: On 06/07/2004, at 10:00 PM, Thomas Niesel wrote: I try the following within context: exten = foo,foo,VoiceMailMain After providung MailBoxNumber I get asked for PassWord. If now the input fails I see on CLI Playing 'vm-incorrect' followed by Playing 'vm-password' and I can hear both messages. Next try is: exten = foo,foo,VoiceMailMain(MBNumber) which jumps straight to the MB and asks my for PW Did you at least configure the mailbox and mailbox password in the mailbox.conf file? Jepp, shure What did you define foo as? it has to be a number Of course, I just post one line from the context. Jean-Yves ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing out of a voicemail message?
Anyway to make hitting `0` during a voice mail dial an extension? The bosses used to have that feature and love it. Their VM prompt would say: Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message. Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc 2 cards working with P2P Mode ?? - massive Problems
I have several machines operating nicely on P2P ISDN lines with QuadBRI's, which uses the same layer2 code... ZapHFC's seem to give a lot of trouble on certain hardware... try using a different machine to host both cards in. Kind regards, Michael Ernst Lehmann wrote: Hello List, is someone operating a DID /P2P / Anlagenanschluss with more than one HFC-Based ISDN-Card ??? I have now 12 hours of setup-troubles behind me with Colt-Telekom, where we did not get it working with two HFC-based cards. Here the setup: - 2 HFC-ISDN-Cards (the one from Conrad-Electronic) - bri-stuff.0.0.2 (with the asterisk-sources from the download.sh-skript) - two NTBAs from Colt-TK with P2P Mode and a block of 30 DID-Numbers The Problem: - when both cards are connected to the NTBAs, and asterisk is started, the Ports in the VST from Colt drop. - when only one card is connected, the Port stays up. - I played with various timing setups... second span with timing=2, or timing=0, but in all cases, the Switch at Colt drops the line... (Layer2-failure they told me) Has anybody a working setup, with two or more HFC-Cards ?? Can you please give me a hint, and which Telco-Carrier you use ?? TIA for all help regarding this. Here the setup: zaptel.conf: span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 zapata.conf: switchtype = euroisdn signalling = bri_cpe pridialplan=local prilocaldialplan=local echocancel=yes echocancelwhenbridged=yes echotrainig=yes overlapdial=no immediate=no group = 3 context=vonaussenkommend channel = 1-2 switchtype = euroisdn signalling = bri_cpe pridialplan=local prilocaldialplan=local echocancel=yes echocancelwhenbridged=yes echotrainig=yes threewaycalling=yes overlapdial=no immediate=no group = 3 context=vonaussenkommend channel = 4-5 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * and Innovaphone
Hi Torsten, I think I have the same problem as Martin Bene mentioned in http://lists.digium.com/pipermail/asterisk-users/2004-January/ 034521.html Since I found no further information about this I'd like to ask wether you know what the reason for this problem is and how one can get around this. I've since spent some time debugging the problem: The innovaphone gatekeeper hands out a bandwidth allocation of 8kbit on registration; I haven't found any way to deactivate or configure this limit. Two possible workarounds: * Don't have asterisk register any extensions with the gatekeeper * Or, as an utterly ugly workaround, I've hacked the openh323 libs to ignore the bandwidth limit and proceed andway. Seems to work OK. Bye, Martin openh323_bandwidth.patch Description: openh323_bandwidth.patch
[Asterisk-Users] SPA-2000 and time of day
Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour slow. I have ntp running on my network and it has been told to respect daylight savings time. Is the SPA omitting this feature? -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do I disable '#' to transfer a call?
Holger Schurig wrote: I don't see anything on the Wiki or in the documentation about disabling this feature. What about the product documentation? Certainly your phone has some means of configuration, e.g. by config files, built-in menus or a web-browser. Use that and the documentation for it. # to transfer is an asterisk feature, not a phone feature. A phone feature would be a button that says Transfer(or some abbreviation/translation thereof.) Look at your Dial line's in asterisk. Do they have a 't' or 'T' in them? - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple E1s over TDMoE?
On Tue, 2004-07-06 at 00:59, Steven Critchfield wrote: Well what is the trouble with moving that information up into variables and using the new functions in IAX to pass that information from one side to the other. Basically, you are going to pass it kind of out of band, but it will get from one side to the other. You can then use it to either place the call or deal with the inbound information. That is a good way of moving information during the call. However, my understanding so far is that information like e.g. the cause of a disconnect could not be transported that way. A call would be hung up by the time the information becomes available. And then, why put work into reinventing the wheel? In theory everything is available in DSS1 and TDMoE could work well as bandwith is not my worry. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-2000 and time of day
David Cook wrote: Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Interesting that it must have broadcast to the local net for a NTP server. From a net admin perspective, I'd consider that a benefit. Q: Where do you tell it to use NTP? snip I didn't see a place either. The net of my problem is that it is 1 hour slow. I have ntp running on my network and it has been told to respect daylight savings time. Is the SPA omitting this feature? Shouldn't the daylight savings be a client configured option? Temporarily Set your NTP server to not be DST friendly and see if the SPA get's the right time. What will happen when the time changes again is so far, undefined. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P PINS
You will have to change signalling to something like a channelized T1 to use a loopback, I think. The PRI has complementary protocols for CPE and NET sides of the link. Not sure if a loopback would come up if it is configured for CPE. -- Steven Critchfield [EMAIL PROTECTED] If I understand what you're trying to do (loop one E1 span to another to see if the card is defective), this should work fine (I've done it). Just loop 1+2 to 4+5, set one span to CPE and the other to NET. You should get green on both. Calls initiated on channels on one will appear as inbound calls on the coresponding channels on the other Regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.723.1 and Asterisk
I have a Cisco ATA 186 working with h323, and G.723.1 codec, but when it makes a connection to a PBX phone, connected to Asterisk by a Digium E100P, don't use G.723.1 codec, the command oh323 show info indicates G.711 for it. Anyone got an idea if Asterisk translates G.723.1 to ISDN channel ? Thanks, Rafael Mayor Rafael Mario Olivieri Comando de Comunicaciones e Informática Dpto Comunicaciones - Jefe Div C4 4346-6137 4346-6100 int 6137 Este mensaje y sus adjuntos son de caracter confidencial para uso de los destinatarios a los que está dirigido. Las opiniones vertidas en este correo son exclusivas de su autor y no representa la opinión del Ejército Argentino. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P problem
Much thanks to those of you who are following this thread. The information has been most helpful. Here's an update for those who are interested. I unplugged everything from every phone line, and tried it all again, and it worked! It worked for about 5 hours. Then, I started to get phantom incoming phone calls that * would answer. The console would indicate that the channel was ringing between 2 and 5 times, and then it would answer, and start going through its speil. As far I I could tell, nothing changed on my end to cause that to happen. I was the only one in the building the entire time, so it's not like someone could have plugged in a phone without me knowing about it. At this point, I gave up working on it, unplugged *, and plugged everything else back in until Monday morning. On Monday morning, I got the same behaviour, with the phantom phone calls. After much troubleshooting, I finally, changed the card out of the machine (I have two cards), made a call out with a regular phone to make sure everything was working properly, rebooted the machine, and once I verified again that there was nothing plugged into the phone line, plugged * back in. At this point, everything worked fine for as long as I was playing with it, which was about 30 minutes. That's where everything stands at the moment. The line that it was plugged into may have been a 4 wire line, so the next thing I'm going to try is to isolate that. I'm at least a little bit comforted that people don't seem to be having these types of problems all over the place. I feel like if I can get it working, then I won't have strange problems after that. Any additional comments or suggestions by people who have had any experience playing with the X100P cards would be most welcome. thanks, Shaun --- Jonathan Biggs [EMAIL PROTECTED] wrote: Just to add some info to this, Hope it will help I had a similar problem when first testing my * setup. I was testing it with an active dual line phone line (all four wires active) and for some reason the X100P did not like that at all. Easiest way is to make sure your line from the jack is just two wires and not a full 4 wire line. Took me several hours to find this one. --- Shaun Dawson [EMAIL PROTECTED] wrote: Hello, folks, I'm having a problem where my X100P isn't behaving the way I think it should. I have the hardware installed fine, with the phone line connected to the port labeled 'line', and nothing in the port labeled 'phone'. The zaptel and wcfxo modules load fine, and there is a line in /var/log/mesages indicating the card if found successfully. I followed directions on the Digium site, which resulted in the following: I added this line to the end of the stock /etc/zaptel.conf file: fxsks=1 I added these lines to the end of the stock /etc/asterisk/zapata.conf file: signalling=fxs_ks context=incoming channel = 1 So far so good. The channel shows up in * in the zap show channels command. I added this to the end of extensions.conf, to make debugging easier: exten = 3000,1,Dial(Zap/1/9728391852) exten = 3000,2,Congestion So, if I dial extension 3000 from a working internal phone, I should dial my cell phone over the zap channel, right? However, if I try to dial out, I connect, but get absolutely nothing. The cell phone doesn't ring, and no audio. If I try to dial in, I get a phone system recording that says due to phone system trouble, this call cannot be completed. This is the type of thing that I'd expect to get if I were plugging in the wrong interface (like if the X100P were for interfacing with phones instead of phone lines), but I've double checked that. Anyone have any ideas? I feel like there is something obvious I'm missing. I have not found anything in the wiki, nor in the mailing list archives. thanks, Shaun __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start.
RE: [Asterisk-Users] TE410P PINS
On Tue, 2004-07-06 at 09:06, Scott Stingel wrote: You will have to change signalling to something like a channelized T1 to use a loopback, I think. The PRI has complementary protocols for CPE and NET sides of the link. Not sure if a loopback would come up if it is configured for CPE. -- Steven Critchfield [EMAIL PROTECTED] If I understand what you're trying to do (loop one E1 span to another to see if the card is defective), this should work fine (I've done it). Just loop 1+2 to 4+5, set one span to CPE and the other to NET. You should get green on both. Calls initiated on channels on one will appear as inbound calls on the coresponding channels on the other This is a wonderful example of many eyes looking at a problem. I completely didn't think about the spare ports. I had tunnel vision on looping the port back on itself. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Termination for Asterisk Users - Inter-Asterisk Exchange
Hi Asha Could you please setup a test account for me and mail me the details thanks Hari"Kanuri, Seshu" [EMAIL PROTECTED] wrote: Folks!Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect.Any volume is good enough for us, even an amount as small as $1.00 a day will do for us.We will provide connectivity from our Softswitch IP 216.162.116.46.If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a test account.My email address is [EMAIL PROTECTED]Thanks and have a great holiday weekendAsha KanuriNetweb Group, Inc.http://www.netwebgroup.com___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone.
Re: [Asterisk-Users] G.723.1 and Asterisk
On Tue, 2004-07-06 at 09:48, [EMAIL PROTECTED] wrote: I have a Cisco ATA 186 working with h323, and G.723.1 codec, but when it makes a connection to a PBX phone, connected to Asterisk by a Digium E100P, don't use G.723.1 codec, the command oh323 show info indicates G.711 for it. Anyone got an idea if Asterisk translates G.723.1 to ISDN channel ? Use google. You will find that the cost of getting G723 implemented in anything is prohibitively expensive due to patents. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
On Tue, 2004-07-06 at 11:29, Martin Bene wrote: The bristuff distribution comes with a install.sh script (./install.sh) which downloads, compiles the required software on your system. If you want to do it manually, look into download.sh to see the exact cvs checkout options which downloads the required asterisk and libpri versions. Yes, I know which libpri/asterisk versions bristuff downloads when using the included scripts (03/24/04). Problem is, I'd like to get the features / bugfixes from later versions. I'd especially like to try current oh323 drives, which require cvs head and don't compile against the versions usd by bristuff 0.2.2. Junghans has promised an update of the software. This was coming 'real soon' (Like when I say - 'I'll be there in 5 minutes' to the wife). I suspect it is even sooner now (promises of this last weekend) - so - sometime soon - and it should work against the current CVS HEAD. -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-2000 and time of day
http://ip/admin/advanced, click on System tab, bottom two options are primary/secondary NTP server. I'm running 2.0.9(d) -Original Message- From: David Cook [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 8:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000 and time of day Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-2000 and time of day
David Cook wrote: Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Interesting that it must have broadcast to the local net for a NTP server. From a net admin perspective, I'd consider that a benefit. Q: Where do you tell it to use NTP? snip I didn't see a place either. It's not uncommon for vendors to embed the IP address of some known time source in code. Use ethereal, reboot the box, and watch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I disable '#' to transfer a call?
On Tue, 2004-07-06 at 01:51, Shaun Ewing wrote: Easy, just don't include t or T in the dial string options. I guess I was searching for the wrong question in the documentation: disabling the transfer feature instead of enabling it. :) I'm only interested in disabling the # when I *make* a call as that's where I'm likely to hit an IVR, so I guess that means removing the 'T' option. Thanks for the help. -- Dameon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000 and time of day
On Tuesday 06 July 2004 17:19, Rich Adamson wrote: It's not uncommon for vendors to embed the IP address of some known time source in code. Use ethereal, reboot the box, and watch. True , and unfortunately, this sometimes goes horrendously wrong... http://www.cs.wisc.edu/~plonka/netgear-sntp/ gdh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New CVS for patch...
Ok, here it goes: I know CVS and I know how to program. I don't know much about linux program installation. I have a WORKING asterisk based on CVS from 04/2004. It's running and, as of three days ago, it's in production as well (production = my wife's using it without knowing it). I want to patch voicemail.c to allow for configurable pager-messages. Looked at the code, and I know I can do that in 10 minutes. Once done, I'm planning to make this patch available to the community, provided the paperwork (release form etc) takes less time than the actual patch. Of course I know that I should based my modification on the latest-available code, but I'm a bit reluctant to upgrade my WORKING asterisk to the latest CVS. Can I rename my asterisk-dir in /usr/src to something different, then check out the latest CVS, make my changes, and if it doesn't work, revert to my working version? Or will Make and its friends throw me for a loop? Thanks -- JM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-2000 and time of day
NTP is time-zone and season agnostic. It always transmits UTC. Offsets from this are set in the client, including DST stuff. If they can't be set, get a better NTP client. :) Chris. David Cook wrote (on Jul 06): Kevin Walsh noted that his SPA-2000 takes time from his local NTP server in a post back on Fri June 25. Q: Where do you tell it to use NTP? I'm a bit confused as to where my SPA-2000 is currently getting its time. I told it GMT-5 in the misc section but it doesn't really tell me where its going for this. Is it just broadcasting looking for ntp? The net of my problem is that it is 1 hour slow. I have ntp running on my network and it has been told to respect daylight savings time. Is the SPA omitting this feature? -- David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How do I disable '#' to transfer a call?
Dameon D. Welch-Abernathy wrote: On Tue, 2004-07-06 at 01:51, Shaun Ewing wrote: Easy, just don't include t or T in the dial string options. I guess I was searching for the wrong question in the documentation: disabling the transfer feature instead of enabling it. :) I'm only interested in disabling the # when I *make* a call as that's where I'm likely to hit an IVR, so I guess that means removing the 'T' option. That means you just want to remove the t/T from your outbound dialplan, not inbound to your extension. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New CVS for patch...
Jay Milk wrote: Of course I know that I should based my modification on the latest-available code, but I'm a bit reluctant to upgrade my WORKING asterisk to the latest CVS. Can I rename my asterisk-dir in /usr/src to something different, then check out the latest CVS, make my changes, and if it doesn't work, revert to my working version? Or will Make and its friends throw me for a loop? You can check out another copy of the CVS code into a directory using the -d parameter to the CVS checkout command. Do your work in there, and produce a patch that can be sent upstream. You'll still have to figure out how to get the patch working with your older version, though, if you want to stick with that version. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New CVS for patch...
On Tue, 2004-07-06 at 10:42, Jay Milk wrote: I want to patch voicemail.c to allow for configurable pager-messages. Looked at the code, and I know I can do that in 10 minutes. Once done, I'm planning to make this patch available to the community, provided the paperwork (release form etc) takes less time than the actual patch. Of course I know that I should based my modification on the latest-available code, but I'm a bit reluctant to upgrade my WORKING asterisk to the latest CVS. Can I rename my asterisk-dir in /usr/src to something different, then check out the latest CVS, make my changes, and if it doesn't work, revert to my working version? Or will Make and its friends throw me for a loop? Not only can you rename your working version, you also are able to checkout to a different directory. Add to it the ability to make you changes to your current install and just backup the modules you are messing with and reinstall it no problem. Then when you are happy with the patch, you could try a test against current to make sure nothing changed in the interim and submit it. The paperwork to allow work to be incorporated into asterisk is basically downloading a form, filling it out(under 2 minutes) and faxing a copy of it and mailing it to Digium. Pretty simple. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New CVS for patch...
Hi Jay, Jay Milk wrote: I want to patch voicemail.c to allow for configurable pager-messages. Looked at the code, and I know I can do that in 10 minutes. Once done, I'm planning to make this patch available to the community, provided the paperwork (release form etc) takes less time than the actual patch. Of course I know that I should based my modification on the latest-available code, but I'm a bit reluctant to upgrade my WORKING asterisk to the latest CVS. Can I rename my asterisk-dir in /usr/src to something different, then check out the latest CVS, make my changes, and if it doesn't work, revert to my working version? Or will Make and its friends throw me for a loop? Yes you can. I do it from time to time. Be sure to remove the contents of /usr/lib/asterisk/modules before installing any version (your current or the latest one), because new modules (if there are any) will not be removed when reverting back to the previous version and you will have problems. And just issue a 'make install' (not a 'make samples'!) -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Randy Bush is a destructive force with a hidden professional agenda
Bradley D. Thornton [EMAIL PROTECTED] wrote: Your on notice as of now and you're being watched. Don't try to destroy this community like the trail of destruction behind you! Who died and made you king of the mail list? From what I can see, Randy Bush asked a question about whether he should use SIP or IAX. It seems that some members of this community are determined to drive him away from IAX and have him use SIP. It would seem to me that anyone new to Asterisk would wonder why a proprietary protocol has been created when SIP is a well established IETF standard. I know the answer, as do many others. If you're unable to post a well reasoned argument for or against IAX then perhaps you should refrain from following up to such questions in the future. People are entitled to ask questions; If no questions were asked then this mail list would not have the volume of articles that it has. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P problem
On Tue, 6 Jul 2004 07:51:53 -0700 (PDT), Shaun Dawson [EMAIL PROTECTED] wrote: On Monday morning, I got the same behaviour, with the phantom phone calls. After much troubleshooting, I finally, changed the card out of the machine (I have two cards), made a call out with a regular phone to make sure everything was working properly, rebooted the machine, and once I verified again that there was nothing plugged into the phone line, plugged * back in. At this point, everything worked fine for as long as I was playing with it, which was about 30 minutes. I've had the same behavior with other phones connected to a X100P line. When someone is using another phone extension, the X100P card becomes confused and continuously goes into my answer context and rings my SIP phones. I havn't tried to debug the problem. I'm at least a little bit comforted that people don't seem to be having these types of problems all over the place. I feel like if I can get it working, then I won't have strange problems after that. Well. I think i'm having the same problem. I've wanted to tweak the zapata.conf to see if I can get rid of it, but I havn't had the time yet. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: is srv lookup being done when REGISTERing?
It's correct that neither the SRV lookup is handled correctly or completely, nore is there in standard distro a way to register with the proxy for a domain, if those names differ. It wasn't a difficult task to change this. If there is interest I might release the patch for this as part of another development. The syntax I implemented was this: register = user[:secret[:authname[:[EMAIL PROTECTED]@proxyhost[:port][/contact] Would this fit your needs? Or any other ideas? There is also the option of expanding, or better redesigning, the [peer] sections with proper and logical configuration options and adding a register=yes flag. Both have been tried, and have their pro's and con's Any thoughts? Senad Jordanovic wrote: Jasminko Mulahusic wrote: This issued has been discussed few weeks ago into great depth. Look into May/June 04 archive! i have indeed looked into archives (wiki, googled using tabs, asked my barber) and what have been discussed were SRV records for outgoing calls. the same seems not to work for REGISTER. Well, I have not followed that thread into great details but my understanding is that * does not support SRV records properly! If you find a concrete answer I would be interested to know about it! BTW... Jasko, gdje se ti nalazis/radis? Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P problem
On Tue, 6 Jul 2004 07:51:53 -0700 (PDT), Shaun Dawson [EMAIL PROTECTED] wrote: On Monday morning, I got the same behaviour, with the phantom phone calls. After much troubleshooting, I finally, changed the card out of the machine (I have two cards), made a call out with a regular phone to make sure everything was working properly, rebooted the machine, and once I verified again that there was nothing plugged into the phone line, plugged * back in. At this point, everything worked fine for as long as I was playing with it, which was about 30 minutes. I've had the same behavior with other phones connected to a X100P line. When someone is using another phone extension, the X100P card becomes confused and continuously goes into my answer context and rings my SIP phones. There was a change submitted via cvs (by Mark) that increased a ring detect parameter about two weeks ago. It got rid of these false rings when using the tdm fxo card, and I'd suspect its the same code for the x100p. If you haven't upgraded lately, might give cvs Head a try. If that doesn't fix it, I'd be looking for noise on the pstn line. Noise can include AC power induction, unbalanced line, etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy running, but moh meetme don't work
Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Response) As well, I set up a meetme conference, and dial it, the first user (also a Sipura sip phone) gets 'there are no other users on the conference.., which is OK, then a second user comes in, but they are not conferenced anymore. I can hang up both phones, and dial back to the conference, but I won't even hear the 'there are no other users message anymore'. usb-uhci and ztdummy are loaded fine (see lsmod), and this system is running Redhat9 standard install with linux sources. Any thoughts what might be wrong? I have already spent the whole night googling and looking around, so I think I covered all the basics already. I tried to use zaptelrtc as an alternative to ztdummy, but it doesn't compile on redhat9 (log below as well), so that is not an alternative either. Is ztdummy fairly reliable, or does it not work on some motherboard usb chipsets? (this is a compaq deskpro pentium 400mhz) Is there something I need to do with my kernel (recompile?) so that ztdummy works, or anything else. (I suspect the cause is ztdummy, since both MOH and Meetme are broken..) Thank you --- Logs/Listings #service zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: wcusb Running ztcfg: [ OK ] #modprobe ztdummy --lsmod listing #lsmod Module Size Used byNot tainted soundcore 6116 0 (autoclean) ztdummy 2532 0 (unused) parport_pc 17508 1 (autoclean) lp 8580 0 (autoclean) parport33952 1 (autoclean) [parport_pc lp] iptable_filter 2316 0 (autoclean) (unused) ip_tables 14488 1 [iptable_filter] autofs 12148 0 (autoclean) (unused) e100 56644 1 wcusb 20064 0 (unused) zaptel179840 4 [ztdummy wcusb] keybdev 2720 0 (unused) mousedev5204 0 hid20772 0 (unused) input 5632 0 [keybdev mousedev hid] usb-uhci 24652 0 [ztdummy] usbcore73088 1 [wcusb hid usb-uhci] ext3 64704 2 jbd47828 2 [ext3] --extensions.conf (relavent part) ;dial 500 to join the conference (doesn't work though) exten=500,1,Answer exten=500,2,MeetMe(1234) ... ;dial 6000 to hear music on hold (doesn't work though) exten = 6000,1,Answer exten = 6000,2,MusicOnHold,default --Meetme.conf [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 --musiconhold.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
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Re: [Asterisk-Users] Wake Up Call AP
Stuart Baggs wrote: Can someone please tell me what sound files to record to get wakeup.agi to work? I'd recommend William Hung's version of She Bangs. If that does not wake up up, nothing will. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: is srv lookup being done when REGISTERing?
Karl Brose [EMAIL PROTECTED] wrote: There is also the option of expanding, or better redesigning, the [peer] sections with proper and logical configuration options and adding a register=yes flag. I would prefer to see a register = yes directive in the type = peer sections of both sip.conf and iax.conf, rather than the current method of using a separate register = whatever directive. The current method could be maintained for backward compatibility, of course. Both have been tried, and have their pro's and con's I see the pros as keeping all of the config in one place, and allowing further options to be added without over-complicating the register syntax. What do you see as the cons? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Randy Bush ?-) intel - telco contract? time will tell.
On Tue, 6 Jul 2004, Kevin Walsh wrote: People are entitled to ask questions; If no questions were asked then this mail list would not have the volume of articles that it has. Absolutly correct - except for Randy who has a tendancy of starting arguments over irrelevant trivia. My own concern is that this list does not degrade into some DNS government mailing list full of trolling. I've seen randy post here many times. Not a problem - but his recent posts smell of trolling. Certainly arrogance and attitude were in abudant evidence. But maybe i'm wrong and just a bit too sensitive based on past experiences. We'll see. One of the things I appreciate in this forum is that people communicate and support each other - and they do a good job of it. We all share the same agenda - bury the telcos and move on. I know randy well - hes intel - would not surprise me if he was here with an agenda on behalf of the spooks he works for and their telco investments. After all he's never participate in anything unless he's gettin paid - randys no fool. but i could be wrong - time will tell. and as far as i'm concerned i really dont' think we should stretch the subject further cheers joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Queue Question
On Sat, 2004-07-03 at 19:12, Sam Tilders wrote:
On Sat, Jul 03, 2004 at 06:45:13PM -0600, Jared Mashburn wrote:
Is there any way for me to add myself to a call queue from outside of my
Asterisk Box?
For example,
I have a queue set up on my asterisk box, and I want to call it on my Cell
Phone, then add myself to the queue and hang up.. When a call comes into the
queue, I want it to be forwarded to my cell phone.
You need to define agents in agents.conf,
agent = agentid,agentpassword,name
where agentid and agentpassword are numbers.
And then add those agents to the queue in queues.conf
member = Agent/agentid
Then in extensions.conf you need:
exten = exten,1,AgentCallBackLogin()
exten = exten,2,Hangup
where the exten number is whatever you dial into asterisk from your
cell.
(You might want to do something like exten/callerid here so
you have some extra validation that it is only you who can call
this extension)
What this then does when you dial it is ask for the agent id, then
the agent password, and then the extension that agent is on.
You would enter an extension that dials your cell phone. You might
need to define one specifically in the same context that uses
your outgoing lines:
exten = cellnumber,1,Dial(${TRUNK}/${EXTEN})
Then when a call is in the queue, it will treat your cell like any
other agent.
Dial the same extension to the callback program to log off by just
pressing # instead of entering an extension.
Is this possible?
I haven't been able to find info on it anywhere, but maybe I'm not looking
in the right help..
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Agents
details a lot of the information you need.
There are some things to be wary of doing this, it seems if no members
handling the queue then the callers will stay there until they give up.
Time periods and phones that are always members can help there.
If anyone knows about how to do proper timeouts when there are no
queue members to call I'd like to hear about it.
--
--
Jared [EMAIL PROTECTED]
Ok, Thanks for the Help, I was able to set this senario up and it seems to work untill
I get to the point where the queue tries to connect it's call to the CellPhone.
My Phone rings but it never creates the bridge... The Debug message says:
Jul 6 10:43:03 DEBUG[409626]: channel.c:2551 ast_channel_bridge: Bridge stops
because we're zombie or need soft hangup:
c0:Local/[EMAIL PROTECTED],2 c1=Local/[EMAIL PROTECTED],1ZOMBIE, flags :
No,No,Yes,Yes
I read somewhere that placing a \n would fix this problem causeing * not to create a
native bridge, but I have had no luck..
How can I fix this problem...?
Thanks Jared
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Re: [Asterisk-Users] T1 configuration, getting help via IRC?
Loopback should always make your status LEDs glow steady green. If that's not working then you've got other problems. It seems I may have those other problems you talked about. I made a loopback cable and tested it on the channel bank. After about three seconds all the status lights went green. I plugged it into the T100P with varying effects. I was grasping a little, and tried different first lines of the /etc/zaptel.conf file: span=1,0,0,esf,b8zs OR span=1,1,0,esf,b8zs cycles between: RED- YEL/RED - YEL/REC - Red/REC - OK. Eventually settles into RED. span=1,0,1,esf,b8zs OR span=1,1,1,esf,b8zs cycles between: RED- YEL/RED - YEL/RED/RED - YEL/REC - RED/REC Any other combinations I tried just came up with a RED alarm. span=1,0,0,esf,b8zs I'd always prefer to clock off of the channel bank; chances are its internal clock is higher stratum than anything on the T100P. :-) If found this heavily commented zaptel.conf: http://www.fnords.org/~eric/asterisk/downloads/zaptel.conf-T100P+PRI If the second number is the timing setting, what number would I put in to get the timing from the channel bank? -paul On another topic, I've tried connecting to irc.freenode.net to join the #asterisk channel. It says I need to be identified to join the channel. pidentd is running on my machine and the firewall is set to allow stuff on 113 through. Anyone else encounter this issue? You need to register with freenode. hop on freenode, then /msg nickserv help for instructions on how to register. bkw_ set the channel mode to require registered accounts to try and stop the spam bots from constantly disrupting the channel. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quantumvoice
http://quantumvoice.com Anybody using this company. They have all you can eat toll free service. Don't see any reference to asterisk, but can use your own Cisco or Sipura. If there is any known working config, appreciate if it could be posted here. DH
[Asterisk-Users] Numbering range
Hi, I found this site to import worldwide number ranges! http://www.numberingplans.com/index.php?goto=isdnaction=analyses=44870 0688688 Does any one know other source(s), preferably free :) Ta Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: is srv lookup being done when REGISTERing?
Kevin Walsh wrote: Karl Brose [EMAIL PROTECTED] wrote: There is also the option of expanding, or better redesigning, the [peer] sections with proper and logical configuration options and adding a register=yes flag. I would prefer to see a register = yes directive in the type = peer sections of both sip.conf and iax.conf, rather than the current method of using a separate register = whatever directive. The current method could be maintained for backward compatibility, of course. I would like to see this implemented as well! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rh9, asterisk HEAD, asterisk-oh323-0.6.3a working
I have no new information, just a note of encouragement to those traversing the bowels of h323: I've been trying to get h323 working with asterisk for several months now, trying with chan_h323 chan_oh323 with all kinds of different combinations. As with several folk on the list, I've had no luck. Either I had no audio, or I could only receive calls, or I could dial but no had no audio. I finally got it working today with the oh323. It seems to work flawlessly in lab settings with both ulaw g729. The command line options for oh323 are odd, but whatever, it works. I'm using RH9 w/ a cvs HEAD checkout from Jul 2 17:54 EST. I would post how I did it, but I only followed the README. -g ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New CVS for patch...
Oh yeah, the -d option. That's what happens if you get pampered by CVS shells all the time. Is there a kind volunteer who'd like to take my updated voicemail.c and perform the needed administrivia? Figuring out the patch-process and disclosure forms is just something I'd rather not do with my current workload. If so, please email me off-list. Thanks! -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 06, 2004 10:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New CVS for patch... Jay Milk wrote: Of course I know that I should based my modification on the latest-available code, but I'm a bit reluctant to upgrade my WORKING asterisk to the latest CVS. Can I rename my asterisk-dir in /usr/src to something different, then check out the latest CVS, make my changes, and if it doesn't work, revert to my working version? Or will Make and its friends throw me for a loop? You can check out another copy of the CVS code into a directory using the -d parameter to the CVS checkout command. Do your work in there, and produce a patch that can be sent upstream. You'll still have to figure out how to get the patch working with your older version, though, if you want to stick with that version. ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ztdummy running, but moh meetme don't work
Make sure you answer the line first. exten = 999,1,Answer exten = 999,2,MusicOnHold(default) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jack Turer Sent: Tuesday, July 06, 2004 11:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ztdummy running, but moh meetme don't work Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Response) As well, I set up a meetme conference, and dial it, the first user (also a Sipura sip phone) gets 'there are no other users on the conference.., which is OK, then a second user comes in, but they are not conferenced anymore. I can hang up both phones, and dial back to the conference, but I won't even hear the 'there are no other users message anymore'. usb-uhci and ztdummy are loaded fine (see lsmod), and this system is running Redhat9 standard install with linux sources. Any thoughts what might be wrong? I have already spent the whole night googling and looking around, so I think I covered all the basics already. I tried to use zaptelrtc as an alternative to ztdummy, but it doesn't compile on redhat9 (log below as well), so that is not an alternative either. Is ztdummy fairly reliable, or does it not work on some motherboard usb chipsets? (this is a compaq deskpro pentium 400mhz) Is there something I need to do with my kernel (recompile?) so that ztdummy works, or anything else. (I suspect the cause is ztdummy, since both MOH and Meetme are broken..) Thank you --- Logs/Listings #service zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: wcusb Running ztcfg: [ OK ] #modprobe ztdummy --lsmod listing #lsmod Module Size Used byNot tainted soundcore 6116 0 (autoclean) ztdummy 2532 0 (unused) parport_pc 17508 1 (autoclean) lp 8580 0 (autoclean) parport33952 1 (autoclean) [parport_pc lp] iptable_filter 2316 0 (autoclean) (unused) ip_tables 14488 1 [iptable_filter] autofs 12148 0 (autoclean) (unused) e100 56644 1 wcusb 20064 0 (unused) zaptel179840 4 [ztdummy wcusb] keybdev 2720 0 (unused) mousedev5204 0 hid20772 0 (unused) input 5632 0 [keybdev mousedev hid] usb-uhci 24652 0 [ztdummy] usbcore73088 1 [wcusb hid usb-uhci] ext3 64704 2 jbd47828 2 [ext3] --extensions.conf (relavent part) ;dial 500 to join the conference (doesn't work though) exten=500,1,Answer exten=500,2,MeetMe(1234) ... ;dial 6000 to hear music on hold (doesn't work though) exten = 6000,1,Answer exten = 6000,2,MusicOnHold,default --Meetme.conf [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 --musiconhold.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: is srv lookup being done when REGISTERing?
And when can we expect a patch from you for this? :P bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Tuesday, July 06, 2004 11:49 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: is srv lookup being done when REGISTERing? Karl Brose [EMAIL PROTECTED] wrote: There is also the option of expanding, or better redesigning, the [peer] sections with proper and logical configuration options and adding a register=yes flag. I would prefer to see a register = yes directive in the type = peer sections of both sip.conf and iax.conf, rather than the current method of using a separate register = whatever directive. The current method could be maintained for backward compatibility, of course. Both have been tried, and have their pro's and con's I see the pros as keeping all of the config in one place, and allowing further options to be added without over-complicating the register syntax. What do you see as the cons? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: is srv lookup being done when REGISTERing?
Senad Jordanovic wrote: Kevin Walsh wrote: Karl Brose [EMAIL PROTECTED] wrote: There is also the option of expanding, or better redesigning, the [peer] sections with proper and logical configuration options and adding a register=yes flag. I would prefer to see a register = yes directive in the type = peer sections of both sip.conf and iax.conf, rather than the current method of using a separate register = whatever directive. The current method could be maintained for backward compatibility, of course. I would like to see this implemented as well! Patches are accepted. Post them on http://bugs.digium.com/ along with a Disclaimer, if you haven't submitted one already. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_cssp problems
Ive seen this question asked a few times but with no resolving answer. Im running CVS-06/24/04-22:20:31, on RedHat Fedora 1, and cannot get chan_cssp to compile. Im getting Now compiling chan_sccp.c 695 lines chan_sccp.c:50: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_sccp.c:50: warning: parameter names (without types) in function declaration chan_sccp.c:50: warning: data definition has no type or storage class chan_sccp.c:51: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_sccp.c:51: warning: parameter names (without types) in function declaration chan_sccp.c:51: warning: data definition has no type or storage class chan_sccp.c:52: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_sccp.c:52: warning: parameter names (without types) in function declaration chan_sccp.c:52: warning: data definition has no type or storage class chan_sccp.c:53: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_sccp.c:53: warning: parameter names (without types) in function declaration chan_sccp.c:53: warning: data definition has no type or storage class chan_sccp.c:58: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_sccp.c:58: warning: parameter names (without types) in function declaration chan_sccp.c:58: warning: data definition has no type or storage class chan_sccp.c: In function `reload_config': chan_sccp.c:421: `devicelock' undeclared (first use in this function) chan_sccp.c:421: (Each undeclared identifier is reported only once chan_sccp.c:421: for each function it appears in.) chan_sccp.c:432: `linelock' undeclared (first use in this function) chan_sccp.c:486: `intercomlock' undeclared (first use in this function) chan_sccp.c: In function `do_monitor': chan_sccp.c:584: `monlock' undeclared (first use in this function) chan_sccp.c: In function `restart_monitor': chan_sccp.c:600: `monlock' undeclared (first use in this function) chan_sccp.c: In function `unload_module': chan_sccp.c:664: `monlock' undeclared (first use in this function) chan_sccp.c: In function `usecount': chan_sccp.c:683: `usecnt_lock' undeclared (first use in this function) make: *** [.tmp/chan_sccp.o] Error 1 Thanks. Harold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy running, but moh meetme don't work
2.4 kernel? I have a RH 9 w/ 2.4 using ztdummy just fine a bit older though. Message seems to show that the phones have trouble reaching each other. Did Sip to Sip between the phones work fine? On Tue, 6 Jul 2004 09:43:18 -0700 (PDT), Jack Turer [EMAIL PROTECTED] wrote: Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Response) As well, I set up a meetme conference, and dial it, the first user (also a Sipura sip phone) gets 'there are no other users on the conference.., which is OK, then a second user comes in, but they are not conferenced anymore. I can hang up both phones, and dial back to the conference, but I won't even hear the 'there are no other users message anymore'. usb-uhci and ztdummy are loaded fine (see lsmod), and this system is running Redhat9 standard install with linux sources. Any thoughts what might be wrong? I have already spent the whole night googling and looking around, so I think I covered all the basics already. I tried to use zaptelrtc as an alternative to ztdummy, but it doesn't compile on redhat9 (log below as well), so that is not an alternative either. Is ztdummy fairly reliable, or does it not work on some motherboard usb chipsets? (this is a compaq deskpro pentium 400mhz) Is there something I need to do with my kernel (recompile?) so that ztdummy works, or anything else. (I suspect the cause is ztdummy, since both MOH and Meetme are broken..) Thank you --- Logs/Listings #service zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: wcusb Running ztcfg: [ OK ] #modprobe ztdummy --lsmod listing #lsmod Module Size Used byNot tainted soundcore 6116 0 (autoclean) ztdummy 2532 0 (unused) parport_pc 17508 1 (autoclean) lp 8580 0 (autoclean) parport33952 1 (autoclean) [parport_pc lp] iptable_filter 2316 0 (autoclean) (unused) ip_tables 14488 1 [iptable_filter] autofs 12148 0 (autoclean) (unused) e100 56644 1 wcusb 20064 0 (unused) zaptel179840 4 [ztdummy wcusb] keybdev 2720 0 (unused) mousedev5204 0 hid20772 0 (unused) input 5632 0 [keybdev mousedev hid] usb-uhci 24652 0 [ztdummy] usbcore73088 1 [wcusb hid usb-uhci] ext3 64704 2 jbd47828 2 [ext3] --extensions.conf (relavent part) ;dial 500 to join the conference (doesn't work though) exten=500,1,Answer exten=500,2,MeetMe(1234) ... ;dial 6000 to hear music on hold (doesn't work though) exten = 6000,1,Answer exten = 6000,2,MusicOnHold,default --Meetme.conf [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 --musiconhold.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: rh9, asterisk HEAD, asterisk-oh323-0.6.3a working
I too had difficulty with chan_h323 driver. However, I used chan_oh323 driver and it worked in the second attempt. The trick is to use the right version on pwlib and openh323 libs. The best way to ensure that is to get them from the same site where you get the chan_oh323 driver. Works like a charm if you just followed the README. If anyone still needs any help, I will be more than willing to do that. BTW, I am also using RH9. -- sudhir ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and H323
Hi guys, I am a newbie in asterisk system. And I wanna to make some questions. I already had a system to solve my VoIP solution, but this system only accept the SIP protocol. Therefore I thinking to using the asterisk like a middle to redirect the H323 calls to my existing system!!! I would like know if the asterisk handle each protocol (SIP and H323) separatedly or if the asterisk translate the protocol?!?! If the first statement is true I can use the asterisk, if not I would like ask if anyone confront a similar problem and what the solutions used. Thanks. Giscard ___ Yahoo! Mail agora com 100MB, anti-spam e antivírus grátis! http://br.info.mail.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
