Re: [Asterisk-Users] iax codec problem
On Sun, Jun 06, 2004 at 04:25:32PM -0400, Tim Sailer wrote: On Tue, Jun 01, 2004 at 07:49:29PM -0500, Yelson Vivas wrote: Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW I have the same problem. IAXCOMM works fine with * 0.7.2, but not 0.9. However, you can make calls fine, just not pick up inbound calls. One workaround is to use Firefly, but that may not be for everyone? To the Firefly maintainer: why does the contacts list fill up with copies of calling parties? Tor ___ 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 Firefly version
On Wed, Jun 02, 2004 at 08:14:44AM +0100, gARetH baBB wrote: On Wed, 2 Jun 2004, Adam Hart wrote: Can I recommend you label files with version numbering - this must be about the third ? fourth ? firefly-thirdparty you've released. .. but have firefly-thirdparty.exe be a symbolic link to the latest version? My 2p. Tor ___ 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] 403 Forbidden between two softphones on same Asterisk
Hi, I have two softphones connected to an Asterisk stable. I have two extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on extension 2000 will ring, but as soon as the call is picked up, extension 2000 will hang up the call. The softphone on 1000 (SIP, SJphone, e.g.) will give a 403 Forbidden result, while a Diax97a on the same extension will just report Call disconnected by remote. The same is not true when 2000 calls extension 1000. Extension 1000 will ring, and is also able to pick up. Extension 2000 can also call external parties (routed through another Asterisk box), but again, external parties cannot call extension 2000 (they can call extension 1000, however!). I'm confident that I've made a mistake, but I just don't know where. Anyone have any ideas? Tor ___ 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: 403 Forbidden between two softphones on same Asterisk
On Wed, Jun 02, 2004 at 11:25:26AM +0200, Tor Houghton wrote:
Hi,
I have two softphones connected to an Asterisk stable. I have two
extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be
completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on
extension 2000 will ring, but as soon as the call is picked up, extension
2000 will hang up the call.
[snip]
I seem to have resolved this problem; for some reason, when upgrading from
an earlier version, the following line was invalid:
exten = 2000,2,Dial(${PHONE1},20,Ttm)
I replaced it with
exten = 2000,2,Dial(${PHONE1},20,t)
And it works fine. I guess I misunderstood the flags during an earlier
configuration of the extensions. Sorry to bother you all.
Tor
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Re: [Asterisk-Users] Problems with IAX Clients, HELP ME PLEASE.
On Wed, Jun 02, 2004 at 05:37:48PM -0300, [EMAIL PROTECTED] wrote: I donwloaded two IAX Clients (firefly and IAX phone) and they did register with *. It would make authenticated calls, but wouldn't actually register with the server. [snip] qualify=1000 i have found that firefly, diax and another iax client (can't remember it now) don't like to be monitored. or something like that. say qualify=no, and it will work. it would be fantastic if it did work. tor ___ 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] MacOS X softphone IAX clients?
Are there any softphone clients that can use IAX/IAX2 for MacOS X? Regards, Tor ___ 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] Free Softphone Recomendations
On Thu, May 20, 2004 at 07:01:09PM +0300, Dan wrote: :-) Dan P.S. You can really decode DTMF tones with your ear/brain?..:-) not far off, but i mostly use the feedback to double check that i didn't dial a number wrong. tor ___ 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] Free Softphone Recomendations
On Wed, May 19, 2004 at 09:16:02AM +0300, Dan wrote: A new version with some cool features (not available on any other soft phone) will be available at the end of the week. Send me a mail if you need further assistance. Looks promising -- one request (I'm sure there will be more); how about DTMF feedback when keying in numbers (e.g. when using voicemail)? Tor ___ 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] Free Softphone Recomendations
On Thu, May 20, 2004 at 12:38:34AM +0300, Dan wrote: Hi Tor, What do you mean by DTMF feedback? When you hit a key, make DIAX play back the corresponding DTMF tone to you. You can enable the key beep in DIAX, but what's the reason to get a DTMF type of feedback? The beep is not enough? For some people, maybe? I just find it more natural to hear the DTMF when I hit a number. It means that if I am dialling a number, I get aural feedback of what I've pressed, which means that I can hear whether or not I've dialled the wrong numbers (e.g. while I am not looking at the screen). Tor ___ 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] 403 Forbidden since upgrading
Hi, I upgraded my local Asterisk (the last version was quite old), and since then, whenever anyone tries to call me via SIP/IAX thru my external Asterisk, they get 403 Forbidden as soon as I pick up. I have no trouble picking up when someone calls via PSTN. Basically, my phone (Firefly softphone) will ring when they call, but will disconnect as soon as I pick up. It won't even go to voicemail, this is from the log on the localiax; -- Accepting unauthenticated call from X.X.X.X, requested format = 4, actual format = 4 -- Executing Ringing([EMAIL PROTECTED]/16385, ) in new stack -- Executing Dial([EMAIL PROTECTED]/16385, IAX2/torh/2201|20|Ttm) in new stack -- Called torh/2201 May 18 18:02:32 WARNING[671760]: chan_iax2.c:2838 iax2_send: timestamp is 0? May 18 18:02:32 WARNING[671760]: channel.c:1445 ast_prod: Prodding channel '[EMAIL PROTECTED]/16385' failed -- Call accepted by 192.168.128.4 (format ULAW) -- Format for call is ULAW -- IAX2[torh]/6 is ringing -- Nobody picked up in 2 ms -- Hungup 'IAX2[torh]/6' -- Executing Hangup([EMAIL PROTECTED]/16385, ) in new stack == Spawn extension (iax, h, 1) exited non-zero on '[EMAIL PROTECTED]/16385' -- Hungup '[EMAIL PROTECTED]/16385' Are there any obvious places I need to look? Tor ___ 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] openbsd compilation fails for recent checkout of v1-0_stable
This has been mentioned before on this list, but in order for md5.c to
compile successfully (OpenBSD 3.3), the following must change in md5.c:
#if defined( __FreeBSD__ ) || defined( __OpenBSD__ )
# include sys/endian.h
Change this to be:
#if defined( __FreeBSD__ ) || defined( __OpenBSD__ )
# include machine/types.h
# include machine/endian.h
And -E is an invalid linker option, so the Makefile needs to be changed:
ifeq (${OSARCH},Darwin)
OBJS+=poll.o dlfcn.o
ASTLINK=-Wl,-dynamic
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
+else
+ifeq (${OSARCH},OpenBSD)
+ASTLINK=-Wl
+SOLINK=-shared -Xlinker -x
else
ASTLINK=-Wl,-E
SOLINK=-shared -Xlinker -x
endif
+endif
Also, for OpenBSD, asterisk's use of gethostbyname_r doesn't work out of the
box, so needs to follow FreeBSD's fixi, by changing the
#if defined(__FreeBSD__)
to
#if defined(__FreeBSD__) || defined (__OpenBSD__)
Also, it is likely that PROC needs to be set manually for your architecture
in the top level Makefile.
Thanks,
Tor
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Re: [Asterisk-Users] sip software
On Wed, Apr 14, 2004 at 02:52:16PM -0400, James Moran wrote: Anyone have any suggestions on free sip phone software for windows?? Only have one IP phone and want to have one other computer hooked up to my Asterisk box for testing. Have you tried Firefly? http://www.virbiage.com/firefly/index.php Supports IAX2 and SIP (though I have not tested it with SIP). Tor ___ 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] Upgraded to latest CVS, now no IAX1?
Hi, I just upgraded to the recent CVS, and IAX1 no longer seems to be available. Is there a way to reenable it? Tor ___ 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] Upgraded to latest CVS, now no IAX1?
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote: Use IAX2, it is a better IAX protocol. Jeremy McNamara P.S. If you really must have it, dig thru the channels/Makefile, but there is zero reason to use it any longer. Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. Tor ___ 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] Upgraded to latest CVS, now no IAX1?
On Tue, Apr 13, 2004 at 04:58:19PM -0400, James Golovich wrote: # If you really want IAX1 uncomment the following, but it is # unmaintained # #CHANNEL_LIBS+=chan_iax.so Thanks all, I'll move to IAX2 after I've tested the notransfer option. Tor ___ 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-to-Asterisk call setup problem (one way works fine)
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote: Tor Houghton wrote: PHONES1=IAX/[EMAIL PROTECTED] Did you try IAX2/[EMAIL PROTECTED] ? Erm, no. Haha, I cannot believe I spent days trying to fix that. It works! My internal asterisk took the call! Yay! Thanks! Tor ___ 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-to-Asterisk call setup problem (one way works fine)
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote: Tor Houghton wrote: PHONES1=IAX/[EMAIL PROTECTED] Did you try IAX2/[EMAIL PROTECTED] ? Actually, I think I found the culprit. It seems (ho hum), that the IAX softphone re-registered (reinvited?) with the external IAX server, so that the translations in the NAT gateway got muddled. (Incidently, do I need to forward udp/5036 and udp/4569 on the NAT gateway when the internal Asterisk registers with the external Asterisk? Or would I only need this if it were the other way round?) Cheers, Tor ___ 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] Asterisk-to-Asterisk call setup problem (one way works fine)
Hi,
I'm having a bit of a problem. I have two Asterisk servers, one serving SIP
clients on the outside of a NAT, the other on the inside. The internal one
also serves PSTN and IAX clients.
When I call someone (who is on SIP) from any phone registered with the
internal Asterisk, I get through to them no problem. The issue is when they
try to call me; for some reason they do not get routed (I am assuming the
extensions are wrong).
The outside Asterisk logs this:
Mar 13 00:18:38 NOTICE[6052352]: app_dial.c:545 dial_exec: Unable to create channel of
type 'IAX'
== Everyone is busy at this time
The outside * extensions.conf contains
PHONES1=IAX/[EMAIL PROTECTED]
[sip]
exten = 2201,1,Ringing
exten = 2201,2,Dial(${PHONES1},20,Ttm)
exten = 2201,4,Hangup
and the outside's iax.conf is so:
[inside]
context=iax
type=friend
secret=PASSWORD
host=dynamic
tos=nodelay
qualify=yes
trunk=yes
Is there something I've missed here? (I'm suspecting I am, but I can't seem
to find any hints on how to fix it.)
Hope someone can help. Been banging my head against this for a few days
without much luck.
Thanks,
Tor
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[Asterisk-Users] OpenBSD patches
Hi, I've applied the OpenBSD patches as noted on http://www.voip-info.org/tiki-index.php?page=Asterisk%20OpenBSD%20patch but there are a few files that still need changing with the current CVS. I've collected them all here (including the ones from the wiki): http://www.bogus.net/~torh/files/asterisk-20040311.patch Of course, I hope these make it into the tree so that OpenBSD users don't have to manually patch + search in future.. :- Tor -- Asterisk CVS-03/11/04-13:23:06 built by [EMAIL PROTECTED] on a i386 running OpenBSD ___ 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] OpenBSD patches
On Thu, Mar 11, 2004 at 11:42:05AM -0600, Tilghman Lesher wrote: On Thursday 11 March 2004 07:45, Tor Houghton wrote: Of course, I hope these make it into the tree so that OpenBSD users don't have to manually patch + search in future.. :- Anything you hope makes it into the tree should be posted to http://bugs.digium.com/ Yeah, John gave me a heads up on that earlier, so I did. Cheers, Tor ___ 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] 2 Linphones communicating through Asterisk?
On Thu, Mar 04, 2004 at 02:06:52PM -, Jon Shamash wrote: [snip] it should be exten = 66,1,Dial(SIP/66) Incidentally, is there a difference between = and =, or are both allowed? Tor ___ 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 channel question
Hi, I've got a question or two about SIP calling channels. As I understand, there is no facility for Asterisk to make outbound calls as if it were a SIP proxy. As I understand it, it is not possible to add an extention that simply states if no match so far, try SIP/url (from what http://www.voip-info.org/wiki-Asterisk+SIP+channels seems to indicate?) Is there a way of passing on such a call, to say a proxy, or should the proxy be the first point of call in such a scenario? I've successfully installed Asterisk on OpenBSD 3.3/x86 (thanks to whoever posted http://www.voip-info.org/tiki-index.php?page=Asterisk+OpenBSD+patch on the wiki), and I am intent on learning more. Regards, Tor ___ 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] Asterisk as proxy?
Hi, So it's like this. I've had siproxd working for me on an external host to which I've established a tunnel (my SIP client is behind a NAT gateway). Of course, I've got to have mailbox functionality at the very least, so a friend of mine told me about Asterisk, which I grabbed from the CVS and installed. However, I've got myself somewhat confused. Do I still tell my SIP client (SJPhone for the time being) to use siproxd as the proxy, or can/should Asterisk be a local (forwarding?) proxy on the NAT side of the tunnel? Basically, the network looks roughly like this, if it helps any: +-+ | siproxd |+ +-+| # | | | T (Internet) U | N | N ++ E | NAT-GW | L ++ | | # ++ ###| VPN-GW | ++ +-+ | +--+ | SJPhone |-+-| Asterisk | +-+ +--+ Hope someone can help. Tor ___ 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
