Re: [Asterisk-Users] What is the best Linux for asterisk
When deciding on Linux you decide which kernel to use. Linux IS the kernel part. After that it's what tools you're most comfortable with. That's where distros vary. In a biz environment you won't probably won't use a GUI. At home (less users) you may want it as a dual function server/ end user pc. So for a most reliable system find the most reliable kernel version. Also, the most reliable version of asterisk would be a more appropriate queston. To sum, there is no magic asterisk linux distro. All have the requisite components at their disposal ( well don't use linspire since they run as root for that ease of use/ hack). On Mon, 2004-08-16 at 09:25, Johannes van Hulst wrote: How has experience in Asterisk voip provider? I am trying to setup a reliable Linux system with Asterisk for a voip provider. Therefore I got two more or like identical systems. System 1 AMD Atlhon XP 2200 Asus A7V600-X bios 1002 1Gb memory 333 Mhz Asus 7100 videocard 120GB harddisk System 2 AMD Atlhon XP 2200 Asus A7V600-X bios 1005 1Gb memory 400Mhz Geforce MX 4000 64MB 40 GB Harddisk At both systems I have problems with installing Linux. I tried Redhat 9.0 but there the systems has badblocks all the time on the ext3 partitions and segmentation errors After that I tried Suse 9.1 and there the system is working perfect only when I compile Asterisk I get compile errors all the time with a warning internal error. I tested the partitions and the memory there is no problem. Can somebody help me out how to get a stabile system? Best regards, Han van Hulst ___ 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 $89 VOIP phone
Only 1 eth port though. The hassle of a full cable run sux sometimes.I don't get making a new phone w/out 2 ports. Makes sense to wait for a more thoughtful design. Although the lcd looks OK. Pluses and minuses. My next phone i want 2 eth, 2 call appearances, and the holy grail IAX ( no more nat issues thank you). My 2 cents on phones. On Mon, 2004-08-16 at 06:00, Holger Schurig wrote: Has anyone tried the new ariavoice $89 VOIP desk phone with Asterisk? ` http://www.voip-info.org/wiki-AriaVoice I am trying it this evening. It is sitting next to my desk, but in white. From what I know so far, this is just another phone based on the PA168 chip from Centrality Comm, so it has it's pro's and con's. For example, the ATCOM AT-323 is very similar. Today I heard from a german reseller of AT323 phones that they now support the IAX protocol. So, put the multiprotocol-capability feature below it's pro-list. ___ 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] Automating calls
/var/spool/asterisk/outgoing On Thu, 2004-06-10 at 15:27, Simon wrote: Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey ___ 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] Small * issue
I've set up a very small * system for a small local paper. The system works great. Here's the issue: I have one of their phone's plugged into the phone port on the x100p and if the phone ring more than 2x then asterisk kicks in and doesn't recognize it as being picked up and starts playing the menu. Can i use wait or something to let the phone ring more and not start the menu? Unfortunately, they are very poor so money for another phone or adapter may not happen. Is a soft phone the only answer? thanks. ___ 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] spandsp w/libtiff-3.6.1?
I had to go use 3.6.0 before it worked. On Mon, 2004-05-31 at 03:08, Aaron J. Angel wrote: Has anyone used spandsp with a patched libtiff 3.6.1 successfully? http://bugs.hylafax.org/bugzilla/show_bug.cgi?id=500 ___ 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 and Zaptel for 2.6 kernel
I'm running it on a Mandrake 10 w/ 2.6, so it should work. On Mon, 2004-05-31 at 13:15, Michael George wrote: We are looking soon at buying a system to deploy asterisk as our company's PBX. We run SuSE here and like it and our asterisk test platform is SuSE 9.0 with the 2.4 kernel. Is anyone running * and the zaptel drivers under SuSE 9.1? With the 2.6 kernel? Is * 64-bit safe (i.e. no 32bit assumptions in the code) so I can run it on an AMD Opteron in 64-bit mode (with whichever kernel is acceptable)? Thank you! -Michael ___ 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] spandsp wont compile.
Yes, success! I deleted the tiff libs I had and installed ver 3.6.0 and was able to compile and load the application modules. Now I just have to do some tweaking and t-shootin' in ext.conf. Thanks and a Shout Out to all for their advice and help. Couldn't have done it w/out you. I also had to put /usr/include in ld.so.conf. Hope this helps others. On Sat, 2004-05-29 at 18:09, Mark Musone wrote: Your most likely compiling against one tiff library version, but loading up another... Do a: ldd app_rxfax.so to see what tiff library it's compiled against, and then also try to find all the places where libtiff is on your machine and remove the incorrect one.. -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Saturday, May 29, 2004 6:09 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] spandsp wont compile. /etc/ld.so.conf /usr/X11R6/lib /usr/lib/qt3/lib /usr/local/libUnable to load module app_rxfax.so May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize /usr/local/lib/libtiff /usr/lib/asterisk/modules the mods compiled BUT now won't load. On Fri, 2004-05-28 at 23:25, Todd Lieberman wrote: add /usr/local/lib to your /etc/ld.so.conf Then run ldconfig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ 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 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 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] spandsp wont compile.
I got it to load BUT now i get when i try to load the module. localhost*CLI load app_rxfax.so localhost*CLI May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize Unable to load module app_rxfax.so May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize On Fri, 2004-05-28 at 22:04, Mark Musone wrote: Make sure that /usr/local/lib is in your /etc/ld.so.conf After you do a make install of spandsp. Also make sure you run ldconfig to update the librarys -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ 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 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] spandsp wont compile.
/etc/ld.so.conf /usr/X11R6/lib /usr/lib/qt3/lib /usr/local/libUnable to load module app_rxfax.so May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource: /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize /usr/local/lib/libtiff /usr/lib/asterisk/modules the mods compiled BUT now won't load. On Fri, 2004-05-28 at 23:25, Todd Lieberman wrote: add /usr/local/lib to your /etc/ld.so.conf Then run ldconfig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vlok Stone Sent: Friday, May 28, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp wont compile. got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ 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 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] spandsp wont compile.
I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ 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] spandsp wont compile.
got it to load but now it errors when starting asterisk. complains of no libspandsp.so.0 and its there. this fax thing is kickin my friggin fax!! On Fri, 2004-05-28 at 13:27, Vlok Stone wrote: I can't get spandsp to compile. when I go to the */apps directory i continually fails. Makefile:80: warning: overriding commands for target `app_rxfax.so' Makefile:77: warning: ignoring old commands for target `app_rxfax.so' cc -fPIC -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [app_rxfax.o] Error 1 I chamged the Makefile to include app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_rxfax.so : app_rxfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_rxfax. o app_rxfax.c app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff app_txfax.o: app_txfax.c gcc -D_GNU_SOURCE -O2 -g -Iinclude -l../include -c -o app_txfax.o app_txfax.c any ideas? thanks in advance. ___ 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] SIP device rings once on busy before giving busy tone with dialplan
here's addition info on sip debug
11 headers, 9 lines
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format telephone-event
Capabilities: us - 14, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: sip:[EMAIL PROTECTED];ftag=as0f38e9f5;lr=on
list_route: hop: sip:[EMAIL PROTECTED]:5028
set_destination: Parsing sip:[EMAIL PROTECTED];ftag=as0f38e9f5;lr=on
for address/port to send to
set_destination: set destination to 192.246.69.223, port 5060
sip show channelsPeer User/ANRCall ID Seq (Tx/Rx)
Lag Jitter Format
192.246.69.223 613 1ecd512b4bf 00103/0 0ms ms
ULAW
192.168.1.247200094915249b0e 00102/01317 0ms ms
ULAW
are these normal?
On Sat, 2004-04-17 at 17:12, Olle E. Johansson wrote:
Chris Orme wrote:
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
Isn't the 'r' forcing a 'ringing' signal from start, regardless
of what the device you are calling are signalling. If you are calling
a SIP device, that device might return 'busy' and that's propably
why you first hear 'ringing' and then a 'busy' signal.
I would like app_dial gurus to explain the 'r' option a bit
more so we can document it better.
/O
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[Asterisk-Users] no sound when connected
I'm having a sound issue. I'm using BT100 (102). When I dial the echo test ( or anything for that matter) outside of my LAN there's no sound when it answers although I hear the ringing tones. Is this an RTP or codec issue. When I dial through Zap everything is fine. Thanx. ___ 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 sound when connected
On Sat, 2004-04-17 at 14:01, Chris Orme wrote: Hi Vlok, When a call connects is the audio one way ? Can the remote person hear you but you can't hear them ? yes. Which way is the audio or is it silent in both directions ? The echo test? Is this FWDs echo test or the one running on your asterisk box (as that is not outside you LAN is it) ? ouside I'm thinking this could be a NAT or firewall issue ? me too. what would i look for. Maybe you could give more info or a diagram of the set up you have there so I can have a think about it? Chris On Sat, 17 Apr 2004, Vlok Stone wrote: I'm having a sound issue. I'm using BT100 (102). When I dial the echo test ( or anything for that matter) outside of my LAN there's no sound when it answers although I hear the ringing tones. Is this an RTP or codec issue. When I dial through Zap everything is fine. Thanx. ___ 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 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 sound when connected
On Sat, 2004-04-17 at 14:43, Chris Orme wrote: Very much sounds like a firewall issue not allowing voice packets back in to you (for the received audio) or them not finding you somehow. Think about how do you connect to the internet. Perhaps 'it' (whatver device it is doing firewalling/NAT) is configurable through its bios or a web interface or by telnet or ssh. Depends what 'it' is, but 'it' is likely to be involved in the problem. You didn't send info of your configuration as to which protocol IAX/SIP you are using and how you are trying to connect so I can't give a specific answer on how to help you. Or I didn't read closely enough. SIP is the protocol. I guess it might be: BT 102 -SIP- Asterisk on local LAN w/PSTN access/zap cards -SIP??- firewall/router -adsl?- internet -SIP- Asterisk (2) yes that's the basic layout. firewall is linux w/ 2 nics iptables are down. I am able to ping out from the asterisk server. So, it is forwarding. iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination still no sound returns from FWD. I'm sure it's the firewall, but can't figure out what's getting denied. But I don't know as you didn't say :-( I know Asterisk went through time when things weren't easy with Grandstream phones, I don't know what the current state of affairs are and I guess it is all great now if it working now via your Zap. www.voip-info.org or using google to search the archives of this list might also help, especially if you search perhaps for the name of your firewall or router and asterisk or something like one way audio? This is what I did when I started. Also if you have available other SIP clients to try on your network and some patience I'm certain this can be tracked down and sorted out. It might even be something as simple as 'nat=yes' 'host=dynamic'. There are lots of sample configs on www.voip-info.org as well as those supplied by Asterisk to work through. Slowly change options from the sample config and with patience you get the hang of things :-) I have nat=yes and host=dynamic Hope that helps a little. Just trying to put something back for all those that helped me. Thank You. You're help is very much appreciated. I hope I may also be of assistance soon to others. Good luck. Chris On Sat, 17 Apr 2004, Vlok Stone wrote: On Sat, 2004-04-17 at 14:01, Chris Orme wrote: Hi Vlok, When a call connects is the audio one way ? Can the remote person hear you but you can't hear them ? yes. Which way is the audio or is it silent in both directions ? The echo test? Is this FWDs echo test or the one running on your asterisk box (as that is not outside you LAN is it) ? ouside I'm thinking this could be a NAT or firewall issue ? me too. what would i look for. Maybe you could give more info or a diagram of the set up you have there so I can have a think about it? Chris On Sat, 17 Apr 2004, Vlok Stone wrote: I'm having a sound issue. I'm using BT100 (102). When I dial the echo test ( or anything for that matter) outside of my LAN there's no sound when it answers although I hear the ringing tones. Is this an RTP or codec issue. When I dial through Zap everything is fine. Thanx. ___ 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 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 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 sound when connected
this is my BT100 phone dtmf. Is this correct.Send DTMF: in-audio via RTP (RFC2833) xvia SIP INFO i chose sip info On Sat, 2004-04-17 at 14:43, Chris Orme wrote: Very much sounds like a firewall issue not allowing voice packets back in to you (for the received audio) or them not finding you somehow. Think about how do you connect to the internet. Perhaps 'it' (whatver device it is doing firewalling/NAT) is configurable through its bios or a web interface or by telnet or ssh. Depends what 'it' is, but 'it' is likely to be involved in the problem. You didn't send info of your configuration as to which protocol IAX/SIP you are using and how you are trying to connect so I can't give a specific answer on how to help you. Or I didn't read closely enough. I guess it might be: BT 102 -SIP- Asterisk on local LAN w/PSTN access/zap cards -SIP??- firewall/router -adsl?- internet -SIP- Asterisk (2) But I don't know as you didn't say :-( I know Asterisk went through time when things weren't easy with Grandstream phones, I don't know what the current state of affairs are and I guess it is all great now if it working now via your Zap. www.voip-info.org or using google to search the archives of this list might also help, especially if you search perhaps for the name of your firewall or router and asterisk or something like one way audio? This is what I did when I started. Also if you have available other SIP clients to try on your network and some patience I'm certain this can be tracked down and sorted out. It might even be something as simple as 'nat=yes' 'host=dynamic'. There are lots of sample configs on www.voip-info.org as well as those supplied by Asterisk to work through. Slowly change options from the sample config and with patience you get the hang of things :-) Hope that helps a little. Just trying to put something back for all those that helped me. Good luck. Chris On Sat, 17 Apr 2004, Vlok Stone wrote: On Sat, 2004-04-17 at 14:01, Chris Orme wrote: Hi Vlok, When a call connects is the audio one way ? Can the remote person hear you but you can't hear them ? yes. Which way is the audio or is it silent in both directions ? The echo test? Is this FWDs echo test or the one running on your asterisk box (as that is not outside you LAN is it) ? ouside I'm thinking this could be a NAT or firewall issue ? me too. what would i look for. Maybe you could give more info or a diagram of the set up you have there so I can have a think about it? Chris On Sat, 17 Apr 2004, Vlok Stone wrote: I'm having a sound issue. I'm using BT100 (102). When I dial the echo test ( or anything for that matter) outside of my LAN there's no sound when it answers although I hear the ringing tones. Is this an RTP or codec issue. When I dial through Zap everything is fine. Thanx. ___ 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 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 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] sound issue
Hi all, I'm new to asterisk. I have it installed and 1 x100p card. I'm trying to use kphone to connect out. But, when I start * it gives me sound card busy error. I've checked ps aux and nothing seems to have the sound card. Any ideas? Does starting asterisk automatically take the sound card? Also, can you use the phone connected into the XFO card w/ asterisk or do you need an XFS card or soft/hard phone? Sorry for the questions, but I can't seem to find any answers on the wiki etc. Thanks in Advance. Vlok ?@ ___ 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
