Re: [asterisk-users] Problems with GXP2000 and Asterisk = Call pickup and Voicemail
Hi thanks for your answer, for dtmfmode, all sip account have dtmfmode=rfc2833 ;=) that's don't change bye Gordon Henderson a écrit : On Fri, 9 Feb 2007, Noc Phibee wrote: Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me [EMAIL PROTECTED] not found .. in features.conf, i have: [general] parkext = 700parkpos = 701-720 context = parkedcalls pickupexten = *8 I'm under the impression that *8 picks up any ringing phone in the same group... Not sure why youre dialling an extension number after it... I may be wrong though - I've never used it! 2- When i want access to the voice server, he never understand my password ... but with a softphone that's work's Anyone have this problems too ? I'd guess that asterisk isn't hearing the tones of the password? Start with putting dtmfmode=rfc2833 in your sip.conf file, and making that setting on the GPX2000 phone itself (on the account page) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error when compiling zaptel-1.4
this is my kernel::: * :/usr/src/zaptel-1.4# uname -r 2.4.27-3-386 also when i type: make clear te rebuild i got errors ** pbx:/usr/src/zaptel-1.4# make clean make[1]: Entering directory `/usr/src/zaptel-1.4/menuselect' rm -f menuselect *.o make[1]: Leaving directory `/usr/src/zaptel-1.4/menuselect' rm -f torisatool makefw tor2fw.h radfw.h rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo make -C wct4xxp clean make[1]: Entering directory `/usr/src/zaptel-1.4/wct4xxp' rm -f *.o rm -f ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_adpcm_chan.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_channel.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_chip_open.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_chip_stats.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_debug.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_interrupts.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_memory.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_mixer.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_phasing_tsst.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_remote_debug.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tlv.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tone_detection.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsi_cnct.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.o ../oct612x/apilib/bt/octapi_bt0.o ../oct612x/apilib/largmath/octapi_largmath.o ../oct612x/apilib/llman/octapi_llman.o make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean make[1]: Entering directory `/usr/src/zaptel-1.4/datamods' make -C /lib/modules/2.4.27-3-386/build SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. make: Entering an unknown directorymake: Leaving an unknown directorymake[2]: *** [archclean] Error 2 make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386' make[1]: *** [clean] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods' make: *** [clean] Error 2 2007/2/8, Yuan LIU [EMAIL PROTECTED]: From: Richard Lyman [EMAIL PROTECTED] Date: Thu, 08 Feb 2007 13:21:58 -0800 when i compile zaptel make linux26 make install i got these errors: make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean make[1]: Entering directory `/usr/src/zaptel-1.4/datamods' make -C /lib/modules/2.4.27-3-386/build SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. Seems to say you don't have full kernel source. That's a requirement for kernel 2.4. Yuan Liu ... i can't believe noone has mentioned he did a 'make linux26' when his kernel is obviously a 2.4 Can't believe myself:-) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone on Linux
On 8 Feb 2007, at 12:33, Tzafrir Cohen wrote: On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote: On 5 Feb 2007, at 21:46, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. You might want to consider our lightweight java softphone (Corraleta SDK) - it can be embedded in a web page - zero install/config in the client. The UI is in HTML and javascript, so you can get it _exactly_ the way you want it. I have a feeling that anything that is written using a Java Plugin will be hevier than a decent Linux desktop program. It's all in the graphics libraries etc. If you are already running firefox, the plugin isn't a huge extra overhead. Xten or Kiax will have a full set of their own .so which almost certainly won't be shared with anything else that is running. The only way to know for sure would be to try it on a sample system - fire up the browser, and click on: http://click.mexuar.com/webuser/click/145/userurl/Westhawk And give me a call (in UK office hours). Zero install would mean Java which is still not exactly zero install in most Linux distributions. It also means that this is not a native applications, and thus has unneeded limitations: you configura the browser and the softphone in two different places. No, not exactly - you configure the softphone on the _server_ all the config is in the surrounding web page (hence on the web server), all your linux images can be identical, and you don't need a (nfs/ samba) fileserver either. (For example: kiax has its own addressboox, but twinkle uses KDE's standard addressbook, which is probably accessible in some other ways). But for a lightweight linux you won't be running KDE :-) Java is a feather in comparison! (I've had corraleta running on a 32Mb 133Mhz arm5 under JamVM, so just 'cos it is Java it doesn't have to be heavy) As for setting it exactly the way you want it: here consider a simple window manager and a very liberal use adaptations per window properties. Yeah, but it is still visibly a softphone, with a web embedded softphone you can make it look and feel like anything your web designer can do. Test if a browser such as dillo or elinks is good enough. If it is: it will save you a whole bunch of memory. And your users will have less to tinker. Consider giving that window a fixed size and location. That's a good option - it all depends on what the user requirements are. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Re: Help - Poor Voice Quality
On 7 Feb 2007, at 16:33, Jim Duda wrote: Tim, What sort of 'poor' quality are we talking about - when folks complain what words do they use? On the other end, folks complain that the voice drops out. Words are lost. It's very frustrating to communicate. Which codec(s) are you using? ULAW How many channels do you want to use at once ? 1 is fine. This is basic home use. What is the round-trip time between you and the teliax server ? The ping responses are on the order of 15mS. I ran mtr, teliax is 10 hops away, and I don't see any packet loss. I was just on the phone with my house, and the call sounded just fine at this time (problems come and go). This is the dump of iax2 show netstats while the call was up. LOCAL - REMOTE ChannelRTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/teliax-2 37 -10-1 -1 0 -1 50 40 0 0 00 0 Do you have the jitterbuffer on or off ? I don't believe so. I didn't turn jitter on. I believe jitter is off by default. Thanks, That's really weird - everything you say makes it look like it would be a great setup, except that it isn't :-( The only thing I can advise now is to get a packet capture (etherreal) of the IAX stream for a bad call and see if the packet timestamps tell us anything. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error when compiling zaptel-1.4
From: younss azzayani [EMAIL PROTECTED] Date: Fri, 9 Feb 2007 08:51:14 + this is my kernel::: * :/usr/src/zaptel-1.4# uname -r 2.4.27-3-386 also when i type: make clear te rebuild i got errors ** ... SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. Kernel 2.4 header will not help you. As mentioned, you need full kernel source with 2.4. Yuan Liu make: Entering an unknown directorymake: Leaving an unknown directorymake[2]: *** [archclean] Error 2 make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386' make[1]: *** [clean] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods' make: *** [clean] Error 2 2007/2/8, Yuan LIU [EMAIL PROTECTED]: From: Richard Lyman [EMAIL PROTECTED] Date: Thu, 08 Feb 2007 13:21:58 -0800 when i compile zaptel make linux26 make install i got these errors: make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean make[1]: Entering directory `/usr/src/zaptel-1.4/datamods' make -C /lib/modules/2.4.27-3-386/build SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. Seems to say you don't have full kernel source. That's a requirement for kernel 2.4. Yuan Liu ... i can't believe noone has mentioned he did a 'make linux26' when his kernel is obviously a 2.4 Can't believe myself:-) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error when compiling zaptel-1.4
On Fri, Feb 09, 2007 at 08:51:14AM +, younss azzayani wrote: this is my kernel::: * :/usr/src/zaptel-1.4# uname -r 2.4.27-3-386 also when i type: make clear te rebuild i got errors ** pbx:/usr/src/zaptel-1.4# make clean make[1]: Entering directory `/usr/src/zaptel-1.4/menuselect' rm -f menuselect *.o make[1]: Leaving directory `/usr/src/zaptel-1.4/menuselect' rm -f torisatool makefw tor2fw.h radfw.h rm -f fxotune fxstest sethdlc-new ztcfg ztdiag ztmonitor ztspeed zttest zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f libtonezone.so libtonezone.a *.lo make -C wct4xxp clean make[1]: Entering directory `/usr/src/zaptel-1.4/wct4xxp' rm -f *.o rm -f ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_adpcm_chan.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_channel.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_chip_open.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_chip_stats.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_debug.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_interrupts.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_memory.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_mixer.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_phasing_tsst.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_remote_debug.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tlv.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tone_detection.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsi_cnct.o ../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.o ../oct612x/apilib/bt/octapi_bt0.o ../oct612x/apilib/largmath/octapi_largmath.o ../oct612x/apilib/llman/octapi_llman.o make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean make[1]: Entering directory `/usr/src/zaptel-1.4/datamods' make -C /lib/modules/2.4.27-3-386/build SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. make: Entering an unknown directorymake: Leaving an unknown directorymake[2]: *** [archclean] Error 2 make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386' make[1]: *** [clean] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods' make: *** [clean] Error 2 I noticed that too. From all I can see in the makefile this shouldn't happen. Anyway, try running 'make' now. Or even 'make clean' again. The last kernel version seems to be remembered somewhere. To reproduce the issue, on a Debian Sarge system with kernel 2.6.8: make clean make KSRC=/lib//modules/2.4.27-3-386/build KVERS=2.4.27-3-386 clean -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Misdn instability with asterisk 1.4
recently i've upgraded asterisk from 1.2.4 to 1.4 All works fine but i'me experencing some instability on misdn channels. In the last week i've experienced twice some problems with misdn (I am using mISDN-1_0_4) dmesg output: mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282 dinfo:] mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282 dinfo:] mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282 dinfo:] mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282 dinfo:] mISDN_rdata: rport queue overflow 256/256 [addr:52020501 prim:120282 dinfo:] What do you think about that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error when compiling zaptel-1.4
On Fri, Feb 09, 2007 at 08:51:14AM +, younss azzayani wrote: this is my kernel::: * :/usr/src/zaptel-1.4# uname -r 2.4.27-3-386 also when i type: make clear te rebuild i got errors ** pbx:/usr/src/zaptel-1.4# make clean [snip] make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean My previous post got it wrong: that line simply shouldn't be there: Index: Makefile === --- Makefile(revision 2130) +++ Makefile(working copy) @@ -506,7 +506,6 @@ $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) clean else $(MAKE) -C wct4xxp clean - $(MAKE) -C datamods clean endif $(MAKE) -C firmware clean rm -f xpp/*.ko xpp/*.mod.c xpp/.*o.cmd Just remove the line marked with a '-' and go on as usual. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error when compiling zaptel-1.4
On Fri, Feb 09, 2007 at 01:34:58AM -0800, Yuan LIU wrote: make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. Kernel 2.4 header will not help you. As mentioned, you need full kernel source with 2.4. My experince shows otherwise. The Debiaan kernel-headers-VERSION (and in Etch, or just about any recent Ubuntu version: linux-headers-VERSION) are good enough to build zaptel. Furthermore, as they install the symlink /lib/modules/VERSION/build that the zaptel makefile uses by default, they are also the easier way to build Zaptel on Debian. This is basically the equivalent of kernel-devel on redhats. Not of the kernel-headers package. The equivalent for that would be kernel-glibc-headers or something similar, that ocntains /usr/include/linux (the stable kernel/userspace API). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone +Realtime
On 7 Feb 2007, at 20:04, Rob Schall wrote: Here's an interesting issue we're facing... We would like users to be able to use softphones from home/work and to use their same extensions they do at work. The first step of getting the phones to log in as their same extensions as work is easy and works. However, on the database side, once the client closes, the sip table is cleared of the ip to the phone. This means that no calls are forwarded to their office line anymore, and instead have to just go to voicemail. To fix this, the best I can think of is to replace those values nightly and update the timestamp so asterisk knows to update its values. Has anyone tried anything like this? I would like the phones to regrab their spot once the softphone is logged out. We have a Asterisk box (gentoo linux) which is running realtime (mysql 5). Our phones are Polycom SoundPoint 501s and the softphone is xlite (windows). I was forced to tackle this a different way - My softphone is IAX and the deskphone is SIP so we can't do SIP tricks. Instead we have separate iax.conf entries for the home phones, and a dialplan for each extension which does checks if the iax channel is available, if it is, the call is routed home, if it isn't the call is routed to the sip desk phone. If you have a consistent naming convention you can get this as a macro. Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best phone for easy provisioning
ci$co phones are definitively not good choice if you would like to use with anything other than callmanager as signaling server (especially true for new models 7911/41/61/70) Michelle Dupuis wrote: We used Aastra's for a good while, but gave up on them (and switched to Cisco). Aastra's seem cheaper up front (hardware costs), but the time wasted chasing firmware bugs, lack of documentation, and poor support quickly eat up any savings. (unless your needs are very basic). MD *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Dovid B *Sent:* Thursday, February 08, 2007 11:21 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Best phone for easy provisioning I liked polycom a lot. - Original Message - *From:* Rod Bacon mailto:[EMAIL PROTECTED] *To:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Sent:* Thursday, February 08, 2007 10:45 AM *Subject:* [asterisk-users] Best phone for easy provisioning Does anyone have any recommendations for a phone that has easy to understand/implement central provisioning? I’ve used CISCO 79XX phones, and they’re great (but too expensive). I like Grandstream phones, but their provisioning sucks. What is everybody else using in large environments where individual config is not an option? Rod Bacon Technical Manager JASCO Consulting Pty. Ltd. http://www.jasco.net.au http://www.jasco.net.au/ Ph. 03 9432 6376 Fax: 03 9432 6378 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] requesting real world meetme capacity numbers
On 9 Feb 2007, at 04:31, JR Richardson wrote: Hi All, I'm very interested in real world experience of double digit number of users sustaining good quality audio in a single meetme conference. Personally, I have seen 23 users in one conf room, all coming in SIP, ULAW. Server is 3.2GHz proc, 1Gig RAM, 1-2 % proc utilization under 23 user load, perfect audio. I'm working on a conf bridge for 150+ users, could use some advice, if anyone has accomplished such a feat or has any ideas on how. I'd be interested in your findings. I'm leaning towards app_conference - you should evaluate it at the very least. With those sorts of numbers you will have to mute the vast majority of users and have them press a button before speaking otherwise the audio will vanish under the weight of 150 people breathing :-( T. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP??
ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729 On Thursday 08 February 2007 19:00, Vicky wrote: config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc ) On 08/02/07, Florea Igor [EMAIL PROTECTED] wrote: Hi, I'm new to *,so i apologize for stupid questions. I'm having problem with this arhitecture: I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm using GSM codec. In extensions.conf I have: exten = 337,1,Dial(SIP/99@ip_pbx2) so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2. RTP stream between sjphone and Asterisk are ok (GSM). The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although ip_pbx2 sip is telling asterisk It only knows codec 0 Is this a config problem or a bug? Igor, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- There are 10 kinds of people in the world: those who know binary and those who don't. Igor Florea Ing. dezvoltare Phone: +40 21 232 04 24 Fax: +40 21 232 31 56 Local time: GMT+2 www.topex.ro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: registration not timing out?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... CLI sip show registry HostUsername Refresh State iinettrunk:5060 [EMAIL PROTECTED] 3584 Request Sent sip.pennytel.com:5060 N 280 Registered Yes, I have same problem. Have you find the solution? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone on Linux
why don't think to sugarcrm, it has an asterisk package, so you benefit of asterisk sugarcrm at the same time Younss AZZAYANI Junior IT Manager Robinson Network ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Billing pulses
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... You then ask the telco to include Advice of Charge (AOC) in your ISDN setup. The AOC then is included somewhere in the Asterisk CDR, but I don't have direct experience of this. You can then get appropriate software to issue bills to telephone users. Unfortunately, as far as I know, Asterisk can't store AOC messages in database. So, provider sends perfectly usable messages, and Asterisk detects them (they are shown on CLI) but it can't store them anywhere. Said. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnection supervision: what about PBX
Because it just works. On 2/8/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Feb 08, 2007 at 01:38:30PM -0500, C F wrote: This device can solve many problems, and is a must for most applications where asterisk is connected using FXO ports and the host PBX deosn't give CPC. http://www.sandman.com/wizard.html#CPCGenerator How does it compare to busydetect of chan_zap ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk and 802.11g
Make sure that your NIC and your X100 are not using the same interrupt. If they are, they will be competing for interrupts and they both will loose. -- -- Steven http://www.glimasoutheast.org Yuan LIU [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I'm greatly surprised when testing an Asterisk box with 802.11g. Here's the topology: VoIP caller --- 802.11g --- Asterisk --- 802.11g --- VoIP extension | FXO ___ PSTN extension When I call a VoIP extension on that box (from a VoIP extension), voice is good. But when this box tries to bridge the call with a PSTN extension, voice is completely broken. And it's not because of the cheap X100P - when I ping the box, round trip is 4,000 ms, most of the time causing timeout. Once the call hangs up, ping time dropped to 1-2 ms. Ping time started to surge even when FXO is simply ringing. If VoIP to VoIP extension call uses re-invite (which it did), voice is also good in the Console channel. How can voice traffic stall 802.11g? (I haven't checked, but CODEC is likely ulaw.) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing pulses
The network terminator installed by the Telco in Romania works the same way: it has two analog outputs and two digital (S0) outputs. I've also got a full TDM400 card with 3 FXS and one FXO, but I gave them up gladly for a proper ISDN card (I'm using a Diva Eicon Server) - and I don't do billing. Sound quality is perfect, there's no echo and I can use all the functions of the ISDN card, like the ability to use multiple MSN's, send an proper busy signal at will, get two calls on the same number at the same time. And now I've got two unused FXS ports in my Asterisk. Stefano Corsi wrote: I must clarify my original message. Maybe confusion is due to my poor english. So I'll make a list of statements: - Each ISDN line in Italy can be splitted in two analog lines - You can use those analog lines as normal analog lines - I have already invested in analog hardware (my fault of course) for both FSX and FXO - ISDN hardware installed by the telco can, in Italy, be programmed to send a billing pulse. - I guess this billing pulse is sent on each of the two analog lines in which a single ISDN line can be splitted (so there's no risk, I guess, for double billing). - I'm considering if there's a small chance for me to avoid buying additional hardware (ISDN cards or gateways) and have an accurate billing using those analog lines resulting from splitting an ISDN line. - To get an accurate billing, I'm wandering if it's possibile to use billing pulse provided by those analog lines. - I have full specifications of the billing pulse provided: frequency 12 kHz ± 1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ± 25 ms pause duration 180 ms period ... 300 ms Do you think it's worth considering it? Rgds Stefano Bill them both. We are talking about mere BRI's, right:-) Good catch, David. As others noted, billing pulse really applies to analogue lines only, and ISDN providers should always send status. Yuan Liu Thanks, Yuan But my confusion came from the original post stating the use of ISDN circuits for this implementation. Id ISDN is in fact the circuit of choice for this app, I agree why wouldn't he simply use the cause codes for billing purposes. We have a lot of experience in telecommunications billing, and have always found cause codes to be more than sufficient even for weird tiers, and bizarre rounding functions. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone on Linux
On Fri, 2007-02-09 at 09:21 +, Tim Panton wrote: On 8 Feb 2007, at 12:33, Tzafrir Cohen wrote: On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote: On 5 Feb 2007, at 21:46, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. [..] It's all in the graphics libraries etc. If you are already running firefox, the plugin isn't a huge extra overhead. Xten or Kiax will have a full set of their own .so which almost certainly won't be shared with anything else that is running. If you are already running firefox give a try to moziax: http://moziax.mozdev.org/ It's a firefox extension for using as IAX2 softphone. MozIAX is free software :) The only way to know for sure would be to try it on a sample system - fire up the browser, and click on: http://click.mexuar.com/webuser/click/145/userurl/Westhawk And give me a call (in UK office hours). [..] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 with 1 FXO
On Fri, Feb 09, 2007 at 01:30:02PM +1100, Klaverstyn, David C wrote: Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling And if I put in rxwink I get this error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring rxwink (Somebody please translate this to asterisk 1.4 syntax) To apply changes in /etc/zaptel.conf, you need to run ztcfg . To apply most changes in /etc/asterisk/zapata.conf you just need to reload asterisk: 'reload' on the CLI, or even more specifically: 'reload chan_zap.so' However certain major changes (basically: adding a channel or chaging its signalling) require a full restart of asterisk: restart now Alternatively, try just 'zap restart' if you have it. It will disconnect existing zaptel calls. You may also need to run it twice. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic Dial, Play message
On Thu, 2007-02-08 at 16:48 -0800, Yuan LIU wrote: From: Stefan Wintermeyer [EMAIL PROTECTED] Date: Thu, 8 Feb 2007 21:56:11 +0100 Am 08.02.2007 um 18:39 schrieb Forrest Beck: Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? Just for example, you have a database of FirstName, LastName, PhoneNumber Jon, Beck, 9194713175 So it would pull each record with phone number, dial the number, when answered play a pre-recorded message. Have a look at an e-mail which I send yesterday to this list. It contains a simple example for a call file. That is the way you want to go. With that you can create a script which solves your problem. Stefan I looked this and http://voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message, both using call files. Can the same commands be used from inside extensions.conf to do same? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The issue is not can you execute the same commands from within extensions.conf, but how are you going to trigger the action without external input. We process calls using the following methodology: 1.Cron starts a job at preset times 2.script log into postgresql and determines if any call are to be made at this time 3.Script then determines how many calls can be made based on codecs, time of day, and service provider to be used 4.Script generates call file/s into temporary directory based on above criteria and moves them to /var/spool/asterisk/outgoing 5.Asterisk places calls, and using cdr_pgsql writes cdr to database 6.upon insert a trigger fires to update list of called numbers and indicate success or failure 7.goto 1 Simple process, extensions.conf is used for all call flow, and no external processes used for updates to database. We used AGi in past and found that this process was actually easier to maintain as the only code written was a simple php script for db access and call file generation. Don't know if this helps with ideas but if you are interested in additional details contact me off list. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnection supervision: what about PBX
On Fri, Feb 09, 2007 at 07:22:27AM -0500, C F wrote: Because it just works. On 2/8/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Feb 08, 2007 at 01:38:30PM -0500, C F wrote: This device can solve many problems, and is a must for most applications where asterisk is connected using FXO ports and the host PBX deosn't give CPC. http://www.sandman.com/wizard.html#CPCGenerator How does it compare to busydetect of chan_zap ? Could you give a scenario where the busydetection of zaptel doesn't work and this one does? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with GXP2000 and Asterisk = Call pickupand Voicemail
1. We just dial the extension directly and have speed dials setup for the first 6 parked positions. We don't use *8 at all. 2. Change the config on the phones under Account to Send DTMF via RTP (RFC2833) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noc Phibee Sent: Thursday, February 08, 2007 9:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problems with GXP2000 and Asterisk = Call pickupand Voicemail Hi i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me [EMAIL PROTECTED] not found .. in features.conf, i have: [general] parkext = 700 parkpos = 701-720 context = parkedcalls pickupexten = *8 2- When i want access to the voice server, he never understand my password ... but with a softphone that's work's Anyone have this problems too ? Thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Any Way to Get # Functionality in DISA
On Thu, 2007-02-08 at 21:48 -0800, Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
Date: Thu, 08 Feb 2007 21:28:03 -0800
Not necessarily. You only have to program your existing context to handle
trailing # when it comes along. For example, this simplistic example
ignores trailing #'s:
exten = _Z., 1, GotoIf($[${EXTEN:-1} = #]?${EXTEN:1},1:2)
exten = _Z., 2, whatever...
Or simply add
exten = _Z.[#*].,1, Goto($[${EXTEN} : \([0-9]*\)],1)
to an existing context to ignore anything after first # or *.
Nope. The . can only be at the end. It matches all remaining chars.
This is not a real fancy pattern matcher. I've seen all sorts of
requests
for a regex there, but the complexity of a regex state-machine in that
code
is staggering!
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Re: [asterisk-users] Queue extension issues
Hello John, I'm not sure - but when tou try to define a context for testq queue with: context=testing it is useless. From what I know you could not have such an option inside a queue. Did you find any documentation specifying a context for a queue? Best regards, ## nini @ www.modulo.ro ## John Breen wrote: I'm stuck on queues! The way I read what documentation I have found, if I set up a queue like this: [general] persistentmembers = yes [testq] musiconhold=default strategy = ringall timeout = 10 retry = 5 context = testing member = SIP/100 and then add into extensions something like this: [incomingiax] exten = 1234,1,Dial(SIP/100,10) exten = 1234,2,Queue(testq|tTH|||300) [testing] exten = 1,1,Dial(SIP/101) [testcontext] exten = 100,1,Dial(SIP/100) exten = 100,hint,SIP/100 exten = 101,1,Dial(SIP/101) exten = 101,hint,SIP/101 exten = 102,1,Dial(SIP/102) exten = 102,hint,SIP/102 Then if a user dials in on extension 1234 (which is what's forwarded from the iax peer), Ext.100 should ring for 10 sec, then the call should be placed in queue testq. While the call is in the queue , the caller should be able to press 1, which should then send them on to ext. 101. That's right, isn't it? Problem is, dialling 1 doesn't go to ext. 101 I can't see anything obvious that I've done wrong - It all looks right to me. But I've obviously missed something. Can anyone enlighten me as to what that something might be? Regards John Breen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording and MWI
Greetings List, I am a newbie and first time mailer so bear with me. I have 2 questions. 1. recording: I have an Meridian Option 11 hooked to my Asterisk box via a PRI with QSIG signalling. I have set up an access code of 8 in the option 11 to access the PRi to the Asterisk Box. Is there a way to set up the record application so that a user could dial 8 and then the number based on the caller id of the user? That is to say if I wanted ext 4711 to be recorded each time it dialed through the asterisk box no matter what number was dialed, how would I set that up? 2. MWI: Has anyone had any experience, or is it possible to send a Message Waiting Indication to an Option 11 over the D channel on the PRI? Thanks for any input you can provide! Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conferencing Phones ...
Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.14 - Chanspy, sound issues.
I upgraded my Asterisk system to version 1.2.14 to check if the sound quality issues I was having with Chanspy in 1.2.7 remained. I'm still getting them, and I'm honestly out of ideas except from RTFS. The called party sounds normally fine, but it's impossible to hear the caller. Sometimes, when the called party is talking, the caller can also be heard. The conversation sounds broken, to the point is almost useless. We don't have any other quality problems beside this. Sound is quite good when making a call or accessing other asterisk services. My setup is as follows: All calls are performed inside a LAN (NOT fully switched...), using SIP and g711. I use SJPhone v1.60 at agents and AT-530 VoIP Phones for the spies. * Intel(R) Pentium(R) 4 CPU 3.00GHz, 1GB RAM, Broadcom Corporation NetXtreme BCM5705_2 Gigabit Ethernet. * Linux foo.bar.com 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux * Asterisk 1.2.14-BRIstuffed-0.3.0-PRE-1w built by bachbuilder @ octopus.physik.fu-berlin.de on a i686 running Linux on 2006-12-19 00:11:55 UTC. Is someone else getting this kind of behaviour? Is Chanspy used normally under this conditions on other installations? Any ideas? saludos, -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy begin:vcard fn:Santiago Aguiar n:Aguiar;Santiago org:;Desarrollo adr:;;Palmar 2548;Montevideo;Montevideo;11600;Uruguay email;internet:[EMAIL PROTECTED] title:NetLabs tel;work:+598 2 7077687 tel;fax:+598 2 7094866 tel;home:+598 2 7075079 tel;cell:+598 99 579739 x-mozilla-html:TRUE url:http://www.netlabs.com.uy/ version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conferencing Phones ...
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, February 09, 2007 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Conferencing Phones ... Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) I have an IP 4000, and I think the quality is excellent (on par with the analogs, which I also consider quite good). Most of our deployments continue to use fxs ports on a channel bank and analog phone, but that's mostly because we have a large investment in them. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] receiving fax with junghanns quadbri bristuff
Hey, anyone know if it's possible to receive faxes through a Junghanns bristuff quadbri card? In germany, currently I have faxes coming in on DID line into QuadBRI and then passing to Digium TDM400 (analog) and into faxmachine. But the reliability of TDM card is spotty, so I want to maybe just accept faxes in on ISDN card and save on asterisk system ...? keeping digital signal strong ... ideas appreciated!! -- -- Chris Earle System Solutions Specialist, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 with 1 FXO
Leo Ann Boon wrote: Klaverstyn, David C wrote: Yes, I have also since put that in and I get the error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring signalling And if I put in rxwink I get this error: Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring rxwink It's all very strange. please post your complete zapata.conf - I think there's a preceding line that's confusing the parser. Leo No, I think what he is doing is a reload, and on a reload Asterisk does not re-setup these settings, so Asterisk is nicely telling you on a reload these are ingnored. I think a 'stop now' would get these settings. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receiving fax with junghanns quadbri bristuff
[EMAIL PROTECTED] schrieb am 09.02.2007 16:12:57: Hey, anyone know if it's possible to receive faxes through a Junghanns bristuff quadbri card? In germany, currently I have faxes coming in on DID line into QuadBRI and then passing to Digium TDM400 (analog) and into faxmachine. But the reliability of TDM card is spotty, so I want to maybe just accept faxes in on ISDN card and save on asterisk system ...? keeping digital signal strong You could receive them with app_rxfax if you are using asterisk 1.2 or another version for which rxfax is still buidable. Otherwise I suggest that you stop using the TDM400 and connect your analogue fax-machine to an S0-Adapter also connected to an quadbri port configured as NT: PSTN - BRI TE - BRI NT - S0-Adapter - analogue fax machine. Works very reliable. cu, Bruno ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] *****SPAMZ***** Conference Page question
Spam detection software, running on the system placebo, has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
[EMAIL PROTECTED] for details.
Content preview: Hi. I'm currently setting up a particular conference: 3
members (a,b,c), a can speak with b and c, b and c can speak only with a
and not between them. I found my possible solution with paging/intercom
using option d (full-duplex), but I need to make ringing the phone in
intercom. Now I set auto-answer=6 but after first ring the phone hangup
the call. There is a way to using page/intercom with normal ring and not
with auto-answer? [...]
Content analysis details: (4.1 points, 4.0 required)
pts rule name description
-- --
5.0 BOTNET Relay might be a spambot or virusbot
[botnet0.7,ip=82.184.107.109,hostname=host109-107-static.184-82-b.business.telecomitalia.it,client,ipinhostname,clientwords]
-1.1 BAYES_05 BODY: Bayesian spam probability is 1 to 5%
[score: 0.0426]
0.0 UPPERCASE_25_50message body is 25-50% uppercase
0.2 AWLAWL: From: address is in the auto white-list
---BeginMessage---
Hi. I'm currently setting up a particular conference: 3 members (a,b,c),
a can speak with b and c, b and c can speak only with a and not between
them.
I found my possible solution with paging/intercom using option d
(full-duplex), but I need to make ringing the phone in intercom.
Now I set auto-answer=6 but after first ring the phone hangup the call.
There is a way to using page/intercom with normal ring and not with
auto-answer?
My dialplan:
[ext-paging]
include = ext-paging-custom
exten = PAGE4441,1,GotoIf($[ ${CALLERID(number)} = 4441 ]?skipself)
exten = PAGE4441,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4441,n,Set(AVAILSTATUS=not checked)
exten = PAGE4441,n,Goto(SKIPCHECK)
exten = PAGE4441,n(AVAIL),ChanIsAvail(${DB(DEVICE/4441/dial)}|js)
exten = PAGE4441,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4441,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4441,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4441,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4441,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4441,n,Dial(${DB(DEVICE/4441/dial)},5, A(beep))
exten = PAGE4441,n(skipself),Noop(Not paging originator)
exten = PAGE4441,n,Hangup
exten = PAGE4441,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = PAGE4442,1,GotoIf($[ ${CALLERID(number)} = 4442 ]?skipself)
exten = PAGE4442,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4442,n,Set(AVAILSTATUS=not checked)
exten = PAGE4442,n,Goto(SKIPCHECK)
exten = PAGE4442,n(AVAIL),ChanIsAvail(${DB(DEVICE/4442/dial)}|js)
exten = PAGE4442,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4442,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4442,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4442,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4442,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4442,n,Dial(${DB(DEVICE/4442/dial)},5, A(beep))
exten = PAGE4442,n(skipself),Noop(Not paging originator)
exten = PAGE4442,n,Hangup
exten = PAGE4442,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
exten = 4446,1,Set(_FORCE_PAGE=1)
exten = 4446,n,Macro(user-callerid,)
exten = 4446,n,Set(TIMEOUT(absolute)=60)
exten = 4446,n,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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Re: [asterisk-users] receiving fax with junghanns quadbri bristuff
Hi Chris, Am 09.02.2007 um 16:12 schrieb Chris Earle: anyone know if it's possible to receive faxes through a Junghanns bristuff quadbri card? In germany So you can read a German documentation? , currently I have faxes coming in on DID line into QuadBRI and then passing to Digium TDM400 (analog) and into faxmachine. But the reliability of TDM card is spotty, so I want to maybe just accept faxes in on ISDN card and save on asterisk system ...? keeping digital signal strong ... Have a look at http://www.das-asterisk-buch.de/stable/faxserver.html Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anyone remembers where to check this list threads on a web site?
Hi all, excuse this doll question, but can´t remember or find where I used to check this list on the web, email is becoming unmanageable along with my regular mail. can anyone provide me withe the link to check the list´s threads under web?:-[ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400: some FXS module fail
Hello, I've installed two Digium TDM2400 cards on my server. One has 24FXS and the other has 16 FXS and 4 FXO. They are both connected to power. Unfortunately some of the FXS module fail to initialize and I find following messages in the logs (the rest of the FXS modules work well). Could someone give me some advice? !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 Feb 9 18:09:45 [kernel] !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 Feb 9 18:09:45 [kernel] !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 Feb 9 18:09:45 [kernel] !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 Feb 9 18:09:45 [kernel] !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 Feb 9 18:09:45 [kernel] !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 4C00 Feb 9 18:09:45 [kernel] !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 Feb 9 18:09:45 [kernel] !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 Feb 9 18:09:45 [kernel] !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 Feb 9 18:09:45 [kernel] !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C Feb 9 18:09:45 [kernel] !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 Feb 9 18:09:46 [kernel] !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 Feb 9 18:09:46 [kernel] !!! CM_BIAS_RINGING iREG 28 = should be C00 Feb 9 18:09:46 [kernel] !!! DCDC_MIN_V iREG 29 = should be C00 Feb 9 18:09:46 [kernel] !!! DCDC_XTRA iREG 2A = should be 1000 Feb 9 18:09:46 [kernel] !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 Feb 9 18:09:46 [kernel] ! Init Indirect Registers UNSUCCESSFULLY. Feb 9 18:09:46 [kernel] Indirect Registers failed verification. Feb 9 18:09:46 [kernel] Port 5: FAILED FXS (FCC) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone remembers where to check this list threads on a web site?
There is a link provided at the bottom of almost every message that will get you close to where you want to be Give it a try. John Novack MF wrote: Hi all, excuse this doll question, but can?t remember or find where I used to check this list on the web, email is becoming unmanageable along with my regular mail. can anyone provide me withe the link to check the list?s threads under web?:-[ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 with 1 FXO
you don't have to connect the power connecter to TDM if you are using FXO, it's used with FXS to generate a signal to phones 2007/2/9, MBIT Technologies [EMAIL PROTECTED]: Hi David Also make sure the power connector is also connected to the board. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 4950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C Sent: Friday, 9 February 2007 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] TDM400 with 1 FXO Hi, Yes it should, I have changed it back and is still causing the same problems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Friday, 9 February 2007 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TDM400 with 1 FXO Klaverstyn, David C wrote: Hi All, I cannot get my TDM to work correctly. In my /etc/zaptel.conf file I have loadzone = us defaultzone=us fxoks=1 Shouldn't this be fxsks if you're using an FXO module as analog trunk? Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conference Page question
Hi. I'm currently setting up a particular conference: 3 members (a,b,c),
a can speak with b and c, b and c can speak only with a and not between
them.
I found my possible solution with paging/intercom using option d
(full-duplex), but I need to make ringing the phone in intercom.
Now I set auto-answer=6 but after first ring the phone hangup the call.
There is a way to using page/intercom with normal ring and not with
auto-answer?
My dialplan:
[ext-paging]
include = ext-paging-custom
exten = PAGE4441,1,GotoIf($[ ${CALLERID(number)} = 4441 ]?skipself)
exten = PAGE4441,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4441,n,Set(AVAILSTATUS=not checked)
exten = PAGE4441,n,Goto(SKIPCHECK)
exten = PAGE4441,n(AVAIL),ChanIsAvail(${DB(DEVICE/4441/dial)}|js)
exten = PAGE4441,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4441,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4441,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4441,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4441,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4441,n,Dial(${DB(DEVICE/4441/dial)},5, A(beep))
exten = PAGE4441,n(skipself),Noop(Not paging originator)
exten = PAGE4441,n,Hangup
exten = PAGE4441,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = PAGE4442,1,GotoIf($[ ${CALLERID(number)} = 4442 ]?skipself)
exten = PAGE4442,n,GotoIf($[ ${FORCE_PAGE} != 1 ]?AVAIL)
exten = PAGE4442,n,Set(AVAILSTATUS=not checked)
exten = PAGE4442,n,Goto(SKIPCHECK)
exten = PAGE4442,n(AVAIL),ChanIsAvail(${DB(DEVICE/4442/dial)}|js)
exten = PAGE4442,n(SKIPCHECK),Noop(Seems to be available (state =
${AVAILSTATUS})
exten = PAGE4442,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=6)
exten = PAGE4442,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE4442,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE4442,n,Set(TIMEOUT(absolute)=60)
exten = PAGE4442,n,Dial(${DB(DEVICE/4442/dial)},5, A(beep))
exten = PAGE4442,n(skipself),Noop(Not paging originator)
exten = PAGE4442,n,Hangup
exten = PAGE4442,AVAIL+101,Noop(Channel ${AVAILCHAN} is not available
(state = ${AVAILSTATUS}))
exten = Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
exten = 4446,1,Set(_FORCE_PAGE=1)
exten = 4446,n,Macro(user-callerid,)
exten = 4446,n,Set(TIMEOUT(absolute)=60)
exten = 4446,n,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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Re: [asterisk-users] SIP??
check your sip.conf and make sure it has allow=ulaw and allow=alaw line ( you can even remove gsm to test it it works fine or not ) On 09/02/07, Florea Igor [EMAIL PROTECTED] wrote: ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729 On Thursday 08 February 2007 19:00, Vicky wrote: config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc ) On 08/02/07, Florea Igor [EMAIL PROTECTED] wrote: Hi, I'm new to *,so i apologize for stupid questions. I'm having problem with this arhitecture: I'm calling asterisk from behind a NAT(sjphone user) with a low band so I'm using GSM codec. In extensions.conf I have: exten = 337,1,Dial(SIP/99@ip_pbx2) so when i dial 337 from sjphone Asterisk is colling 99 on ip_pbx2. RTP stream between sjphone and Asterisk are ok (GSM). The problem is rtp packets from Asterisk to ip_pbx2 are also GSM although ip_pbx2 sip is telling asterisk It only knows codec 0 Is this a config problem or a bug? Igor, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- There are 10 kinds of people in the world: those who know binary and those who don't. Igor Florea Ing. dezvoltare Phone: +40 21 232 04 24 Fax: +40 21 232 31 56 Local time: GMT+2 www.topex.ro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chan_Cellphone
Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BindPort
The point is to use more than one port, I think the only way is to use the redirect from iptables On 2/6/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Ciao, just change port value in sip.conf. Giorgio Il Neofita wrote: Hi, I was wondering if it is possible to set asterisk in order to listen to different ports for the sip or I need to do this operation with iptables? All of this since some time the port 5060 is blocked. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conferencing Phones ...
We use the Polycom soundstation 2W plugged into an iaxy...works very well... Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com [EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 - Fax. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, February 09, 2007 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Conferencing Phones ... Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dependencies on DB?
What are the specific dependencies that Asterisk has on databases? Some hi-perf data is stored in BDB, CDRs are in a relational DB like MySQL. Is there a list of specific dependencies by specific modules on specific tables? A complete list, so switching from the default DB can drop the old DB from the install. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chan_Cellphone
This isn't included in the trunk for the moment. You have to use the patch to get chan_cellphone. Regards, Tristan Mahé Il Neofita a écrit : Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.2.13 released!
The Asterisk development team is pleased to announce the release of Zaptel 1.2.13. This release contains a large number of bug fixes, an important performance improvement for most Digium cards, and support for new Digium hardware and some significant improvements in the XPP driver for Xorcom's Astribank hardware. In detail: * A modification was made to the drivers for all Digium PCI cards to improve their compatibility and performance when used in interrupt sharing environments. * Support for the Digium TDM800P 8-port analog interface card was added. * Support for the Digium TC400B 92/96-channel transcoder card was added. * Support for the Digium High Performance Echo Canceller add-on software module was added. * All drivers updated to Linux kernel 2.6.20 API changes. * Performance improvements for multiple Astribank units. * Astribank firmware protocol version is now 2.4. * Astribank now supports Message Waiting light on analog telephone sets. * Added a /proc interface to blink the leds on the Astribank to identify ports in large setups. * fxotune is now supported by Astribank. All users of Zaptel 1.2.x are encouraged to update to this release as soon as they can practically do so. Thanks for your support of Asterisk and Zaptel! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.15 released!
The Asterisk development team is pleased to announce the release of Asterisk 1.2.15. This release contains a large number of bug fixes, and some significant improvements: * Support for Zaptel-based transcoder hardware, initially the Digium TC400B 92/96 channel transcoder. * Handling of voicemail subdirectories when using ODBC storage has been improved, so that messages can be forwarded properly. * A problem with forwarding voicemails from folders other than the user's INBOX has been fixed. * The Zaptel channel driver can now support echo cancellers that provide 64ms or 128ms of echo cancellation per channel. Thanks for your support of Asterisk and Zaptel! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Any Way to Get # Functionality in DISA
From: Steve Murphy [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 07:11:50 -0700
On Thu, 2007-02-08 at 21:48 -0800, Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
Date: Thu, 08 Feb 2007 21:28:03 -0800
Not necessarily. You only have to program your existing context to
handle
trailing # when it comes along. For example, this simplistic example
ignores trailing #'s:
exten = _Z., 1, GotoIf($[${EXTEN:-1} = #]?${EXTEN:1},1:2)
exten = _Z., 2, whatever...
Or simply add
exten = _Z.[#*].,1, Goto($[${EXTEN} : \([0-9]*\)],1)
to an existing context to ignore anything after first # or *.
Nope. The . can only be at the end. It matches all remaining chars.
This is not a real fancy pattern matcher. I've seen all sorts of requests
for a regex there, but the complexity of a regex state-machine in that
code is staggering!
Thanks for pointing out, Steve - just spent two sleepless hours swallowing
my own pill:-) Well Asterisk has one regex engine already, then that's
alright.
Yuan Liu
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[asterisk-users] Re: Chan_Cellphone
I start the patch and automatically created the file. But now on the menu I cannot select chan_cellphone I launched ./bootstrap.sh and after ./configure in my /usr/include/bluetooth I have the header but I cannot select the option any idea? On 2/9/07, Il Neofita [EMAIL PROTECTED] wrote: Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] After upgrade to 1.4 transfers don't workproperly
Ok that worked for normal transfers. Now here is another situation. When we try
to transfer a call directly to voicemail it plays the voicemail message but we
can't transfer the call. The only way I could get it to work was to do a
conference and then drop out of that conference.
My dial plan for direct dialing is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
When this is attempted the following message shows up on the CLI of Asterisk:
[Feb 9 11:52:03] WARNING[5054]: chan_sip.c:12310 handle_response: Notify
answer on an owned channel?
Can anyone tell me what this means and what I can do to fix this?
Thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote:
I have discovered an issue on my system after upgrading from 1.2.13 to
1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone.
I have confirmed this on multiple phones. When the called person
answers and tries to transfer the call to another extension, the call
successfully transfers, however the person answering the transfer
cannot hear the person that called in, the caller. My dial command
simply is
I had exactly the same problem when upgrading to 1.4 and I solved by
making sure canreinvite=no is in sip.conf for every phone.
--
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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[asterisk-users] Re: asterisk-users Digest, Vol 31, Issue 37
On Friday 09 February 2007 11:50, [EMAIL PROTECTED] wrote: Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) Cheers, Gordon The quality of a conference phone is determined more by how it's designed and manufactured than whether it uses VoIP or analog. We've deployed a couple of IP4000s and they work great. The nice thing about them is that if you already have a bunch of SoundPoint IP phones, they require nothing special in regards to provisioning since they use the same firmware and configuration as the rest of the SoundPoint IP series. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Way to Get # Functionality in DISA
Yuan LIU wrote:
From: Steve Murphy [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 07:11:50 -0700
On Thu, 2007-02-08 at 21:48 -0800, Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
Date: Thu, 08 Feb 2007 21:28:03 -0800
Not necessarily. You only have to program your existing context to
handle
trailing # when it comes along. For example, this simplistic example
ignores trailing #'s:
exten = _Z., 1, GotoIf($[${EXTEN:-1} = #]?${EXTEN:1},1:2)
exten = _Z., 2, whatever...
Or simply add
exten = _Z.[#*].,1, Goto($[${EXTEN} : \([0-9]*\)],1)
to an existing context to ignore anything after first # or *.
Nope. The . can only be at the end. It matches all remaining chars.
This is not a real fancy pattern matcher. I've seen all sorts of requests
for a regex there, but the complexity of a regex state-machine in that
code is staggering!
Thanks for pointing out, Steve - just spent two sleepless hours
swallowing my own pill:-) Well Asterisk has one regex engine already,
then that's alright.
DISA is used to emulate picking up an analog phone and dialing out. If
you don't want to emulate that, then use something else like Read().
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Re: [asterisk-users] BindPort
I also encountered the problem of port 5060 being blocked by some user's isp and redirected port 5098 to 5060 but still asterisk wasnt able to detect hangup properly and had load of voice problems ( lot of nat involved and softphones were being used ) so i made asterisk listen on 5098 and redirected port 5060 to 5098 via iptables and it solved all problems ( port block users were able to use 5098 completely while other users had no problems with 5060 too ) . Try this method if u get some voice problems for port blocked users . On 09/02/07, Il Neofita [EMAIL PROTECTED] wrote: The point is to use more than one port, I think the only way is to use the redirect from iptables On 2/6/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Ciao, just change port value in sip.conf . Giorgio Il Neofita wrote: Hi, I was wondering if it is possible to set asterisk in order to listen to different ports for the sip or I need to do this operation with iptables? All of this since some time the port 5060 is blocked. Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux Kernel Timer Frequency and Asterisk
1000 Hz is recommended if you use lot of meetme channels ( and maybe iax trunking ? ) without a hardware timer . On 08/02/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Wed, 7 Feb 2007, Mark Coccimiglio wrote: Ok here is a real geek question, I building my own linux kernel for my asterisk system and came across the kernel setting for the timer frequency. I have one of 3 hardcode choices 100Hz, 250 Hz and 1000Hz. From what I understand the default Freq was changed from 100Hz in kernel 2.4 to 1000Hz (1KHz) in kernel 2.6. Timing is a BIG issue in asterisk with all the TDM and zap channel stuff. My guess is to go with the lower 100 or 250 Hz option but that is only a guess. The 1KHz sounds like it will conflict with the Zap 1khz timer (or am I wrong about that). Does anyone know what the prefered settings are for Trixbox or AsteriskNOW (or the asterisk code forks e.g. OpenPBX)? Please let me know what your experience has been. I always compile custom kernels and have been using 1KHz in all my systems which are for anything vaguely interactive. Most of my asterisk systems are 1GHz processors but I have a small handful which are 533MHz and all are working just fine. You're not using a recently kernel then ;-) 2.6.20 offers 300Hz too (supposedly good for video applications) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID on Dish 301 Receiver
I have a Dish 301 receiver that will not display CallerID when connected to FXS module on TDM400. Uniden phone connected to the same FXS module does display CallerID. When Dish 301 receiver is connected to IAXy CallerID is displayed properly. Any suggestions on getting the CallerID to display on the Dish 301 receiver through the TDM400 FXS module? Hugh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RFC2833 SIP trunks and DTMF
I have a telco providing DTMF inband, they say they can't provide it any other way. This is creating headaches for me. What is the common method for SIP DTMF? Kpml, or 2833 or inband? My handsets don't support inband so I'm tying up some expensive resources to convert the inband DTMF to out-of-band DTMF... Can you recommend a vendor in US that provides SIP with DTMF in RFC 2833? - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 FC5 and Gtalk
JABBER: gtalk_account OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to=' gmail.com' version='1.0' localhost*CLI jabber show tes JABBER: gtalk_account INCOMING: ?xml version=1.0 encoding=UTF-8?stream:stream from=gmail.com id=58D5EEFB06C20E13 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client [Feb 9 21:11:15] ERROR[2061]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. I installed all the gnutls but I still have this error [EMAIL PROTECTED] ~]# rpm -qa | grep gnutls gnutls-utils-1.2.10-3 gnutls-1.2.10-3 gnutls-devel-1.2.10-3 Do you know how to solve it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detect hang-up
I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect hang-up
On Fri, 2007-02-09 at 15:31 -0500, David Ruggles wrote: I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! Maybe my zapata.conf can help you. I've one X100P working for almost 2 years :) [channels] language=es context=from-pstn signalling=fxs_ks rxwink=300 ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=no hidecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no relaxdtmf=yes inmediate=yes busydetect=yes busycount=6 callprogress=yes musiconhold=default echotraining=400 rxgain=-4.0 txgain=4.0 group=0 callgroup=1 pickupgroup=1 TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ring requested on channel
Hi guys i have a problem with an isdn (E1) pri works fine but once or twice a week i got ring requested on channel X then every channel get blocked so i should restart the pbx to fix it, i try not using cdr mysql, several linux distros and every 1.2.x asterisk version, even i try to ask (sensei Mark still waiting ) the weird thing is that i have the same problem with three more servers connected to the same telco and the same local exchange (ericsson) So any hint more that welcome Thanks Br Yelson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: Friday, February 09, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Detect hang-up On Fri, 2007-02-09 at 15:31 -0500, David Ruggles wrote: I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! Maybe my zapata.conf can help you. I've one X100P working for almost 2 years :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
I've been doing some googling and I found references to using debug=1 with wctdm to see what's actually going on. It says this will be printed to the console. I'm running my * box headless in another room and sshing in to the box. I can't find where the debug out (if there is any) is going. Can any one point me in the right directions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 09, 2007 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Detect hang-up Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receiving fax with junghanns quadbri bristuff
On 10:12, Fri 09 Feb 07, Chris Earle wrote: Hey, anyone know if it's possible to receive faxes through a Junghanns bristuff quadbri card? In germany, currently I have faxes coming in on DID line into QuadBRI and then passing to Digium TDM400 (analog) and into faxmachine. But the reliability of TDM card is spotty, so I want to maybe just accept faxes in on ISDN card and save on asterisk system ...? keeping digital signal strong ... Can you repost this to the bristuff-users list? http://lists.three-dimensional.net/mailman/listinfo/bristuff-users Thanks -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4 FC5 and Gtalk
Ciao Neofita. I'm trying my GTalk account and I'm still having the same problem. I've installed the gnuTLS-developer rpms and rebuilt and re-installed the complete Asterisk package but without success. I'm working with OpenSuse 10.2. This is my debug info that's quite similar to what you've posted: JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='gmail.com' version='1.0' Gateway*CLI JABBER: asterisk INCOMING: ?xml version=1.0 encoding=UTF-8?stream:stream from=gmail.com id=2601C222D846D6C3 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttls xmlns=urn:ietf:params:xml:ns:xmpp-tls/mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanism/mechanisms/stream:features [Feb 9 23:15:43] ERROR[24214]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. There is someone knowing what's the problem and that could help us? Best regards, Marco Signorini. -- Passa a Infostrada. ADSL e Telefono senza limiti e senza canone Telecom http://click.libero.it/infostrada9feb07 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
Awww... This is when I feel stupid, and for the sake of others... I will expose my shame: Be sure you run `autoconf` after applying the patch (and making the required changes to configure.ac) Since it's altering configure.ac afterall, and not configure; then of course run configure and etc. I did so and now it works... except I now have the disappointment of realizing that ulaw over SIP isn't really all that well suited to fax. (hey... 1/3 of a page is better than nothing... r-right?) Anthony Kepler wrote: Did you ever find a solution for this? I'm in the same boat with 1.4.0-beta3 and SpanDSP - Anthony Kepler Matt Gibson wrote: Okay, So, More updates after testing some more 1. with the free line commented out of app_rxfax.c, and recompiled, asterisk seems to work on non-fax incoming calls to my fax extension. Doesn't send a file obviously, but does seem to actually reach the right place and do what it's supposed to do. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
From: David Ruggles [EMAIL PROTECTED] Date: Fri, 9 Feb 2007 16:43:41 -0500 I've been doing some googling and I found references to using debug=1 with wctdm to see what's actually going on. It says this will be printed to the console. I'm running my * box headless in another room and sshing in to the box. I can't find where the debug out (if there is any) is going. Can any one point me in the right directions? Two things to try. One is to simply start in console mode remotely (forget safe_asterisk). The other is to modify safe_asterisk script and disable console on ttyS9. Then when you start a remote console, STDERR will be there. Yuan Liu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 09, 2007 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Detect hang-up Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
By your post I can conclude that the console wctdm debugs to is the asterisk console. In that case I'm not getting anything from wctdm. I'm not using the safe_asterisk script I'm running asterisk -cvvv from the command line. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Friday, February 09, 2007 5:45 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Detect hang-up From: David Ruggles [EMAIL PROTECTED] Date: Fri, 9 Feb 2007 16:43:41 -0500 I've been doing some googling and I found references to using debug=1 with wctdm to see what's actually going on. It says this will be printed to the console. I'm running my * box headless in another room and sshing in to the box. I can't find where the debug out (if there is any) is going. Can any one point me in the right directions? Two things to try. One is to simply start in console mode remotely (forget safe_asterisk). The other is to modify safe_asterisk script and disable console on ttyS9. Then when you start a remote console, STDERR will be there. Yuan Liu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 09, 2007 4:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Detect hang-up Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Detect hang-up
From: David Ruggles [EMAIL PROTECTED] Date: Fri, 9 Feb 2007 16:23:18 -0500 Thanks for the conf file, but it didn't make any difference. If I hang-up during a record it will hang the channel until I stop Asterisk. If I hang-up during playback I get the following: [Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension 'D', but no rule 'i' in context 'incoming' If this offers a clue. Search recent and past archive for disconnect supervision and so on. Bottom line is, you need telco to offer certain capability. Because you use an analogue line, my impression is that Kewl start is common place in North America, but some still use loop start. Some send you an audible fast busy tone to indicate disconnect. I got one line like this. So I enabled callprogress=yes. It helps, but not very reliable. If they don't even send a tone, you are stuck. As to Invalid extension 'D', you'll need to pose relavant portions of your dial plan to determine if it is a real clue. Yuan Liu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: Friday, February 09, 2007 3:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Detect hang-up On Fri, 2007-02-09 at 15:31 -0500, David Ruggles wrote: I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! Maybe my zapata.conf can help you. I've one X100P working for almost 2 years :) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue extension issues
Ioan Indreias wrote: Hello John, I'm not sure - but when tou try to define a context for testq queue with: context=testing it is useless. From what I know you could not have such an option inside a queue. Did you find any documentation specifying a context for a queue? Best regards, ## nini @ www.modulo.ro ## Indeed I did, in several places including on the web. One I believe was in the excellent book from O'Reilly Asterisk - The Future of Telephony, Also from the sample configs with * 1.2.14, which is what we're using: ; A context may be specified, in which if the user types a SINGLE ; digit extension while they are in the queue, they will be taken out ; of the queue and sent to that extension in this context. ; ;context = qoutcon Regards John Breen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan checkup
Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten = s,1,Wait(1) exten = s,2,IfFax continue to ring, so that the Fax Machine gets it exten = s,3,Answer exten = s,4,Playback(Message) exten = s,5,Dial(SIP/1000SIP/2000SIP/3000) exten = s,6,Hangup() exten = fax,1,Wait(30) exten = fax,2,Wait(10) exten = fax,3,Hangup() I'm wanting the line to ring, If it is a fax coming in then Asterisk leaves the line alone and lets the fax machine handle the call. If it is a call then Asterisk answers, plays a greeting and rings the IP phones? Question is how does asterisk detect the call without answering? I'm not wanting Asterisk to handle the call if it is a fax if possible??? I look forward to your input, Thank You Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with GXP2000 and Asterisk = Call pickupand Voicemail
Ken Williams wrote: i have two problems with my Grandstream GXP2000 : 1- When i wan pickup a call, that's don't work's (*8EXTEN) and when i test whit Softphone, i have a error too, he say me [EMAIL PROTECTED] not found .. in features.conf, i have: *8 doesn't take an extension does it? If you dial *8 send, it just picks up the first ringing extension within your pickup group. At least, that's how I use it on several sites... Regards John Breen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan checkup
On Fri, 2007-02-09 at 18:35 -0500, Barry Fawthrop wrote: Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten = s,1,Wait(1) exten = s,2,IfFax continue to ring, so that the Fax Machine gets it exten = s,3,Answer exten = s,4,Playback(Message) exten = s,5,Dial(SIP/1000SIP/2000SIP/3000) exten = s,6,Hangup() exten = fax,1,Wait(30) exten = fax,2,Wait(10) exten = fax,3,Hangup() I'm wanting the line to ring, If it is a fax coming in then Asterisk leaves the line alone and lets the fax machine handle the call. If it is a call then Asterisk answers, plays a greeting and rings the IP phones? First, the in s,3 you answered the line. If the fax machine doesn't answer on the first ring, it never will, because once asterisk picks up the line, there won't be any more ringing. Second, when s comes to an end, it will listen for a response, and be able to hear and respond to the fax tone. In this case, you hang up before that happens. So the fax extension can't be activated. Question is how does asterisk detect the call without answering? I'm not wanting Asterisk to handle the call if it is a fax if possible??? I look forward to your input, Thank You Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dialplan checkup
From: Barry Fawthrop [EMAIL PROTECTED] Date: Fri, 09 Feb 2007 18:35:43 -0500 Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten = s,1,Wait(1) exten = s,2,IfFax continue to ring, so that the Fax Machine gets it exten = s,3,Answer exten = s,4,Playback(Message) exten = s,5,Dial(SIP/1000SIP/2000SIP/3000) exten = s,6,Hangup() exten = fax,1,Wait(30) exten = fax,2,Wait(10) exten = fax,3,Hangup() I'm wanting the line to ring, If it is a fax coming in then Asterisk leaves the line alone and lets the fax machine handle the call. If it is a call then Asterisk answers, plays a greeting and rings the IP phones? Question is how does asterisk detect the call without answering? I'm not wanting Asterisk to handle the call if it is a fax if possible??? My impression is no (reliable) way. Asterisk doesn't do silent answer kind of stuff. However, if you can be flexible, many FAX machines have silent answer and includes a feature to ring the hand set if it determines the call is not for FAX. You may be able to find a model that allows ringing of an external phone set. Yuan Liu I look forward to your input, Thank You Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outbound Call Transfer Problem
Hi
I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per call.
The problem happens:
- With both software and hardware phones.
- With calls going out through the ZAP channel and to internal SIP
extensions.
- After I have transferred an incoming call once, I can not transfer it
again.
My features.conf looks like:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls ; Which context parked calls are in
parkingtime = 240 ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout = 3 ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
;adsipark = yes ; if you want ADSI parking announcements
;pickupexten = *8 ; Configure the pickup extension.
Default is *8
featuredigittimeout = 1000
[featuremap]
blindxfer = #
atxfer = *
sip.conf snippet:
[603]
type=friend ; This device takes and makes calls
username=603; Username on device
secret=hlpme2go ; Password for device
canreinvite=no
host=dynamic ; This host is not on the same IP addr every time
context=from-sip ; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
; voicemailbox has messages in it
And an abridged extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
include = parkedcalls
exten = _6XX,1,Dial(SIP/${EXTEN},30,T)
exten = _6XX,2,Voicemail(u${EXTEN})
exten = _6XX,102,Voicemail(b${EXTEN})
exten = _6XX,103,Hangup
exten = _04.,1,Macro(dial-mobile,${EXTEN})
[macro-dial-mobile]
exten = s,1,SetGlobalVar(NumToDial=${ARG1})
exten = s,2,SetGlobalVar(theCHANNEL=ZAP/3)
exten = s,3,Dial(${theCHANNEL}/${NumToDial},60,T)
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s,104,Goto(s-CHANUNAVAIL,1)
exten = s-CHANUNAVAIL,1,SetGlobalVar(theCHANNEL=SIP/iinet)
exten = s-CHANUNAVAIL,2,Playback(voip-warning)
exten = s-CHANUNAVAIL,3,Dial(${theCHANNEL}/${NumToDial},60,T)
exten = s-CHANUNAVAIL,4,Hangup
; zap/3 context
[home]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,Wait(1)
exten = s,6,Dial(SIP/600SIP/601SIP/602SIP/603,75,t)
Any suggestions?
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[asterisk-users] asterisk and multiple cpus/cores
I have found a site that list the following (no date in the post, so it may be old): since all transcoding and calls still go through one core in asterisk, it doesn't make sense to buy a multi-core or hyperthreaded system that will only slow you down Does that still applies in asterisk 1.2.14/1.4.x ? Or do we have to tweak source code to balance loads (transcoding,etc) between cores? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The High Performance Echo Canceller (HPEC)
I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? TDM800P and HPEC The TDM800P is an 8-port analog telephony interface card, so it fills the gap between Digium's 4-port and 24-port cards. Analog phones and POTS lines are going to be with us for some time, and demand for support for them remains high. The TDM800P is a bus-mastered PCI card, which means it installs in legacy hardware and provides better performance than CPU-controlled cards. The High Performance Echo Canceller (HPEC) is a software upgrade to legacy Digium cards, and is included with the new TDM800P. The HPEC is supposed to be the greatest thing since Lydia Pinkham's Vegetable Compound, and surefire cure for echo problems. It is host-based, so it's not dependent on the interface card. It's free to Digium customers, and available at $10 per channel for non-Digium cards. http://www.voipplanet.com/trends/article.php/3657981 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect hang-up
On Fri, Feb 09, 2007 at 02:45:17PM -0800, Yuan LIU wrote: From: David Ruggles [EMAIL PROTECTED] Date: Fri, 9 Feb 2007 16:43:41 -0500 I've been doing some googling and I found references to using debug=1 with wctdm to see what's actually going on. It says this will be printed to the console. I'm running my * box headless in another room and sshing in to the box. I can't find where the debug out (if there is any) is going. Can any one point me in the right directions? Two things to try. One is to simply start in console mode remotely (forget safe_asterisk). The other is to modify safe_asterisk script and disable console on ttyS9. Then when you start a remote console, STDERR will be there. You seem to confuse several things: The messages that were enabled are debug messages emmited by a kernel module. Messages of a kernel modules may be printed to the console (depending on the console logging priority. You normally *don't* want it to print debug messages directly to the console. You can also see the recent kernel messages (a buffer of at least 16kB of the last mkernel messages) as the output of 'dmesg' . Kernel messages are also sent to syslog. Most systems are configured not to log debug messages. But you can edit /etc/syslog.conf (and restart syslog) to have it to make it log debug messages, or all kernel messages, or whatever. For instance, the following line in syslog.conf: kern.* /var/log/kern.log Will log all kernel messages (including debugging ones) to /var/log/kern.log. To look at the latest: tail -f /var/log/kern.log (I'm not aware of a simple way of doing this with dmesg) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] changing callerid to ring groups callerid
Hi all! First off all, sorry for my bad english. I have a setup where some of the users have several extensions(work, home, mobile etc). Therefore i have made a ring group for each of the users with more than one extension. The ring group is set up to use ring all. What i want is that no mather what extension a user calls from, I want the Ring-Group number to be the callerid. That way the other user only have to remember one number for each of the other users, even though they might have several extensions. Anyone? -- Best regards Bjorn Marius ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: asterisk 1.4 FC5 and Gtalk
i saw the same problem and here is a thread where i mentioned how i fixed it.. http://lists.digium.com/pipermail/asterisk-users/2006-November/171783.html look for my previous mails in this thread sometime september-november 2006 . btw, i can't get asterisk to work with google talk yet. thanks sridhar From: [EMAIL PROTECTED] Reply-To: asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 31, Issue 39 Date: Fri, 9 Feb 2007 18:30:38 -0700 (MST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by bay0-mc11-f20.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Fri, 9 Feb 2007 17:36:29 -0800 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTP id 1C63E2FCC32;Fri, 9 Feb 2007 18:30:38 -0700 (MST) X-Message-Info: txF49lGdW41RYrP+Tdoh49JHbOTLdMhagyFw1S6VRR0= X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 10 Feb 2007 01:36:30.0423 (UTC) FILETIME=[E3926E70:01C74CB3] Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Message: 12 Date: Fri, 9 Feb 2007 23:28:15 +0100 From: marcotasto [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk 1.4 FC5 and Gtalk To: asterisk-users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Ciao Neofita. I'm trying my GTalk account and I'm still having the same problem. I've installed the gnuTLS-developer rpms and rebuilt and re-installed the complete Asterisk package but without success. I'm working with OpenSuse 10.2. This is my debug info that's quite similar to what you've posted: JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='gmail.com' version='1.0' Gateway*CLI JABBER: asterisk INCOMING: ?xml version=1.0 encoding=UTF-8?stream:stream from=gmail.com id=2601C222D846D6C3 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:clientstream:featuresstarttls xmlns=urn:ietf:params:xml:ns:xmpp-tls/mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanism/mechanisms/stream:features [Feb 9 23:15:43] ERROR[24214]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. There is someone knowing what's the problem and that could help us? Best regards, Marco Signorini. -- Passa a Infostrada. ADSL e Telefono senza limiti e senza canone Telecom http://click.libero.it/infostrada9feb07 _ Gossips, movie reviews, photogallery and more http://content.msn.co.in/Entertainment/Default ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan checkup
Thanks Guys I already have the fax machine a brother all-in-one Printer, scanner, fax. I realize the s,3, answers the line But How can I get s,2, to detect if it is a fax and take it from there without answering? Or can someone explain what make an incoming goto exten = s,.. and what make it go to exten = fax, How does this logic work?? Thanks again Barry Barry Fawthrop wrote: Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten = s,1,Wait(1) exten = s,2,IfFax continue to ring, so that the Fax Machine gets it exten = s,3,Answer exten = s,4,Playback(Message) exten = s,5,Dial(SIP/1000SIP/2000SIP/3000) exten = s,6,Hangup() exten = fax,1,Wait(30) exten = fax,2,Wait(10) exten = fax,3,Hangup() I'm wanting the line to ring, If it is a fax coming in then Asterisk leaves the line alone and lets the fax machine handle the call. If it is a call then Asterisk answers, plays a greeting and rings the IP phones? Question is how does asterisk detect the call without answering? I'm not wanting Asterisk to handle the call if it is a fax if possible??? I look forward to your input, Thank You Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and multiple cpus/cores
Erick Perez wrote: I have found a site that list the following (no date in the post, so it may be old): since all transcoding and calls still go through one core in asterisk, it doesn't make sense to buy a multi-core or hyperthreaded system that will only slow you down Does that still applies in asterisk 1.2.14/1.4.x ? Or do we have to tweak source code to balance loads (transcoding,etc) between cores? I can tell you that statement is bogus. We run a number of dual cpu and single cpu systems on our network. The dual ones (Xeon 3.6Ghz) can easily handle 90 G729 calls at 50% CPU Usage. The single ones will be at 50% with only 40 calls. Andres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan checkup
From: Barry Fawthrop [EMAIL PROTECTED] Date: Fri, 09 Feb 2007 21:49:17 -0500 Thanks Guys I already have the fax machine a brother all-in-one Printer, scanner, fax. I realize the s,3, answers the line But How can I get s,2, to detect if it is a fax and take it from there without answering? It is not about dial plan. Your original plan should work if you can get Asterisk/Zaptel to do silent answer (in priority 2). Otherwise nothing is going to work. On the other hand, all Brother IntelliFAX support silent answer (something they call intelligent ring or the like). See if your machine has an TAD (telephone answering device) port - and old Brother I used had this. If yes, you can enable silent answer on the FAX, then connect your X100P/Asterisk to this TAD port. FAX machines usually monitors for FAX tone for 2 rings. Some machines also cut off ring to TAD before it determines it the call is not for FAX; if not, simply program your Asterisk to wait for a few seconds before doing anything. This should give you the same functionality. Yuan Liu Or can someone explain what make an incoming goto exten = s,.. and what make it go to exten = fax, How does this logic work?? Thanks again Barry Barry Fawthrop wrote: Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten = s,1,Wait(1) exten = s,2,IfFax continue to ring, so that the Fax Machine gets it exten = s,3,Answer exten = s,4,Playback(Message) exten = s,5,Dial(SIP/1000SIP/2000SIP/3000) exten = s,6,Hangup() exten = fax,1,Wait(30) exten = fax,2,Wait(10) exten = fax,3,Hangup() I'm wanting the line to ring, If it is a fax coming in then Asterisk leaves the line alone and lets the fax machine handle the call. If it is a call then Asterisk answers, plays a greeting and rings the IP phones? Question is how does asterisk detect the call without answering? I'm not wanting Asterisk to handle the call if it is a fax if possible??? I look forward to your input, Thank You Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call park and call transfer example
Hi Ango - Does any can give me some example to setup call parking and call transfer of a call? In my understanding, call parking and call transfer should be like something below. Am I right? Call parking: caller A - callee B callee B park her call callee B get back her call in another phone Call transfer: caller A - callee B callee B transfer to C finally: A talks to C Yeah, that more or less describes parking and transferring. Parking and asterisk-based transfers are set up in features.conf. I'll suggest you have a look at the sample features.conf file, or you can look at the wiki, too: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound Call Transfer Problem
I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. Any suggestions? I have questions: 1) what version of 1.2? 2) Anything come up in the CLI? How about the logs? Have you tried turning on verbose logging in logger.conf? (be sure to turn it off when you're done) 3) What SIP phones are you using (hard and soft)? - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call park and call transfer example
Noah, Thanks for you reply. I have a problem in call parking as following. scenario 1 1.Caller A - callee B 2.Callee B answered 3.callee B dial # to park the call and hear transfer 4.callee B dial 700 to park the call 5.callee B hang up and caller A hear 701 Why caller A hear the call parked number 701 instead of callee B? scenario 2 1.caller A - callee B 2. callee B answered 3. caller A dial # to park the call and hear transfer 4. caller A dial 700 to park the call 5. caller A hear 701 and hangup I think scenario 2 is more reasonable compared with scenario 1. I wonder whether callee can park the call. Any comment? ango On 2/10/07, Noah Miller [EMAIL PROTECTED] wrote: Hi Ango - Does any can give me some example to setup call parking and call transfer of a call? In my understanding, call parking and call transfer should be like something below. Am I right? Call parking: caller A - callee B callee B park her call callee B get back her call in another phone Call transfer: caller A - callee B callee B transfer to C finally: A talks to C Yeah, that more or less describes parking and transferring. Parking and asterisk-based transfers are set up in features.conf. I'll suggest you have a look at the sample features.conf file, or you can look at the wiki, too: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound Call Transfer Problem
Noah Miller wrote: I am using Asterisk 1.2 and for the life of me, I am unable to transfer outbound calls (eg calls I initiate from sip extensions). When I press #, nothing happens. Inbound calls transfer fine, but only once per call. Any suggestions? I have questions: 1) what version of 1.2? version 1.2.1 2) Anything come up in the CLI? How about the logs? Have you tried turning on verbose logging in logger.conf? (be sure to turn it off when you're done) nothing at all. 3) What SIP phones are you using (hard and soft)? sipura 2000 and sjphone. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP retry time too low
I have a problem with asterisk-1.2.13, where it retries SIP INVITEs too quickly. It happens when qualify is on, and the server it tries to reach is only 1ms away according to qualify. The time between the first SIP INVITE and the 7th (last) is then only 64ms, and that can be too short for the peer to react. I reported this bug in much more detail in bugs.digium.com, but the bug is gone now without even an email saying where it went. I don't remember the issue number. Somewhat frustrating. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
