Re: [asterisk-users] VoIP GSM Gateways
On 18:15, Sat 28 Oct 06, Forum wrote: I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Hi, We are using a voiceblue in our office. It's a voiceblue that can take 2 sims but we only use one. It's working great and we have no issues with it. Junghanns.net also has a pci card with 1, 2 or 4 simslots. That looks very good but I have no experience with it. -- 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] Would you support a Bristuff mailing list ?
Hi,Up to now, I can think of 6 people claming interest in such list.As bristuff somehow modifies Asterisk, people using bristuff and asking support in Asterisk User list will be forwarded to this list.So this number should be growing. Maybe, a core of 20 people would be enough to turn this idea into something successful.I think we can easily reach this target.So ? ___ --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] app_meetme not loading
I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for timing. I have now installed the zaptel package and I believe the ztdummy module is loading ok [EMAIL PROTECTED] asterisk-1.4.0-beta2]# lsmodModule Size Used byztdummy 5672 0zaptel 207908 9 ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 2497 1 zaptelI have tried to recompile asterisk again by doing the followingmake cleanmakemake installHowever Asterisk still does not compile app_meetme. Is there soemthing else I should be doing? regardsWil ___ --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] How to make different ext using different trunks?
Put them in diffrent context's and then have a seperate context allowing everyone to talk to each other (locally) and have an include in every context. - Original Message - From: Zeeshan Zakaria To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, October 29, 2006 5:17 AM Subject: [asterisk-users] How to make different ext using different trunks? Hi, I want to do so that extension 501 will always use trunk1 for outbound calls and 502 will use trunk2 for outboud calls. How do I do this. Right now all extensions use the same trunk for outbound calls.-- Zeeshan A Zakaria ___ --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] Diva server 4bri and Portuguese BRI
Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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 I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.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
Re: [asterisk-users] Compiling Zaptel 1.2.10 on Ubuntu 6.10
Dear Storm, I have two guesses One could be something in the ubuntu make which makes it unable to understand some regx in the scripts used or I am not quite sure but check the kernel version you are having (i do that by uname -a ) I believe you will find something there, if it is not the same as the one in ubuntu 6.06 then try installing the kernel of 6.06 else I have no idea. Thx' MAG Strom Carlson wrote: Here's a weird problem that I'm not quite sure how to resolve. Zaptel 1.2.10 compiles just fine with "make", but when "make install" is run, this happens: [ `id -u` = 0 ] /sbin/depmod -a 2.6.17-10-generic || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 "" "tor2 torisa wcusb wcfxo wctdm wctdm24xxp ztdynamic ztd-eth wct1xxp wcte11xp pciradio ztd-loc ztdummy" [: 66: ==: unexpected operator [: 66: ==: unexpected operator Unknown kernel build version requested... exiting. make: *** [install] Error 1 This worked just fine under ubuntu 6.06 with the same set of packages installed. Any help is appreciated. -- Strom Carlson http://www.stromcarlson.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 -- Thx MAG ___ --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: [OT] wi-fi ip phone scenario
Martin Joseph wrote: I think it's cleary true that wiring WIFI infrastructure is easier and more reliable then WDS. On the other hand, I have been running my little network with WDS for over three weeks now, and it has been completely reliable. The tricks where to configure things properly and to have the bases closer together then one would think would be needed. Once this was setup. It works, and it keeps working. We had a couple of stress tests also, one black out and one unplugged router (carpenter). Came up cleanly and continued working fine. No mis-registrations and no problems. Marty Can I ask you guys which phones are you using? Alberto. ___ --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] translate.c:88 powerof: Powerof 0: No power??
2006/10/28, Moises Silva [EMAIL PROTECTED]: This is a problem of the codec you are attempting to use. Wich codecis?, it seems to Asterisk cannot identify the codec you are using.RegardsOn 10/28/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi,I have just installed fresh FreeBSD 6.1 and asterisk. But I get warning about power. I have read forums that others had the same problems, but there is no solution how to fix this problem. Maybe somebody knows how to fix this problem... This is my machine configurations: KERNEL: options IPFIREWALL options DUMMYNET options HZ=1000 and commented these: #cpuI486_CPU #cpuI586_CPU#optionsINET6 I have installed asterisk from ports. This is my installed programs for asterisk: mpg123-0.59r_17 asterisk-1.2.13 asterisk-addons-1.2.3_1 spandsp-0.0.2.p26 zaptel-1.0_1 libpri-1.2.3 Loaded modules: Id Refs AddressSize Name19 0xc040 66a2a4 kernel21 0xc0a6b000 58554 acpi.ko32 0xc367f000 31000zaptel.ko41 0xc36b4000 2000 ztdummy.ko 51 0xc36cd000 16000linux.ko Warnings: Oct 28 16:38:35 WARNING[429]: translate.c :88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c :88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:88 powerof: Powerof 0: No power?? Oct 28 16:38:35 WARNING[429]: translate.c:133 ast_translator_build_path: No translator path from unknown to unknown ___ --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 --Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org___--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 I'm just using SIP. In general parameters I wrote:disallow=allallow=alawallow=ulawAnd in users the same. ___ --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 behind NAT and without portforwarding forrtp
Am Sunday 29 October 2006 01:31 schrieb Dovid B: Half asleep. Sorry for my last post. I believe you still need port forwarding for IAX. Time to keep to my bed time. If works as long as you have notransfer=no at both ends. Iam concerned that with SIP Asterisk is bridging up and I do not receive the audio stream. Asterisk should Hangup the line if Audio stream is announced to com from another IP. Iam wonderung that there is no setting for this. ___ --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: Voicemail and OSX 10.4 Intel
Well, yes I'm sorry I'm using 1.2.13, all compile is ok, also I've installed mpg123. all modules loaded fine, and all codecs too. One thing keep my attention and is that when I change format for recording, it ever uses wav|wav49, I tried to change on voicemail.conf format to only gsm, but is not recording in this format. format=gsm 2006/10/27, Martin Joseph [EMAIL PROTECTED]: On 2006-10-27 09:59:10 -0700, David Parcerisa [EMAIL PROTECTED] said: Hello; I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac Pro intel box. When I try to record a message from an incoming call or a greeting message from internal phone using voicemail, It's like something is not doing well. I can heard anything, only a distorsion sound that is equal to lenght of the message left. First I thoug that could be something with format=gsm|wav. I think tha could be something related to this : x=0, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav49, 0x518fe0 -- x=1, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav, 0x180a200 but I don't know what this means ... something I need to compile extra? thanyou in advance Why are you using 1.2.1? try updating to something a bit fresher like 1.2.12.1;~) I have never seen any issue with this on my mac asterisk systems so I don't think it's something extra to build. You should see these in your /usr/lib/asterisk/modules by default. Did you mess around with your module loading or your modules? You might have screwed things up that way... Dunno really, just reaching, Marty ___ --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] app_meetme not loading
On Sun, Oct 29, 2006 at 04:47:48PM +0800, Will Roy wrote: I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for timing. I have now installed the zaptel package and I believe the ztdummy module is loading ok [EMAIL PROTECTED] asterisk-1.4.0-beta2]# lsmod Module Size Used by ztdummy 5672 0 zaptel207908 9 ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 2497 1 zaptel I have tried to recompile asterisk again by doing the following make clean make make install However Asterisk still does not compile app_meetme. Is there soemthing else I should be doing? What version of zaptel do you have? Asterisk 1.4 requires zaptel 1.4 to build chan_zap. (or a small tweak to zaptel.h and some symlinks to provide the new locations, if you don't really want to bother your system with zaptel 1.4. Though zaptel 1.4 should hopefully be fully compatible with older chan_zap versions). -- 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] Pager Voicemail Message
Hello,In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system.Is there a way to manipulate this message, as well?Thanks,David___ --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] porting numbers in UK telewest/bt/adept
On 26 Oct 2006, at 11:59, Conrad Wood wrote: A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. What is the old PBX and how are Telewest presenting? We had Telewest lines once and they were the same RJ-45 ISDN 30 as BT. Would it not be possible to use a dual port card and use Adept for the outgoing and Telewest for the incoming service? -- Matthew Thompson [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] VoIP GSM Gateways
Hi, On Sun, 2006-10-29 at 09:16 +0100, Michiel van Baak wrote: On 18:15, Sat 28 Oct 06, Forum wrote: I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Junghanns.net also has a pci card with 1, 2 or 4 simslots. That looks very good but I have no experience with it. Also we have a 2/4 gsm channels card. Many thing is that is not zaptel based and do not require any asterisk patching. Please take a look to our wiki, http://open.voismart.it were full docs are hosted. greetings, Matteo ___ --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] VoIP GSM Gateways
On Sun, Oct 29, 2006 at 01:27:25PM +0100, Matteo Brancaleoni wrote: Hi, On Sun, 2006-10-29 at 09:16 +0100, Michiel van Baak wrote: On 18:15, Sat 28 Oct 06, Forum wrote: I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. What experience have people on this list have with GSM gateway hardware. I have been looking at the 2N voiceblue products. Junghanns.net also has a pci card with 1, 2 or 4 simslots. That looks very good but I have no experience with it. Also we have a 2/4 gsm channels card. Many thing is that is not zaptel based and do not require any asterisk patching. Please take a look to our wiki, http://open.voismart.it were full docs are hosted. Is vISDN (extra kernel modules, extra non-standard Asterisk channel) required? The page on vGSM there suggests it is. -- 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] VoIP GSM Gateways
2006/10/29, Forum [EMAIL PROTECTED]: Peter,How much does the 4 port cost? How many simultaneous calls can you make? Doyou need a mobile account from a mobile provider such as T-mobile?I've been told it costs 1600 Euros for 4 ports and 1400 for 2 ports. Regards ___ --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] Diva server 4bri and Portuguese BRI
Thanks Alberto! I tested with telsampl like you said (with various configurations for de diva) and this not works...:( The trace is: Enter destination address: 273xx --Conn_Req(273xx) Connect_Con-- [29]:Disc_Ind-- --Disc_Res **Call cleared*** Any idea for the possible problem? Thanks and best regards, PS. 2006/10/29, Alberto Pastore [EMAIL PROTECTED]: Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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 I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.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 ___ --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] Diva server 4bri and Portuguese BRI
pls post your misdn.conf as well as extensions.confMay be i can help.Sou Português:)On 10/29/06, Pedro Silva [EMAIL PROTECTED] wrote:Thanks Alberto!I tested with telsampl like you said (with various configurations for de diva) and this not works...:(The trace is:Enter destination address: 273xx--Conn_Req(273xx)Connect_Con--[29]:Disc_IndDisc_Res**Call cleared*** Any idea for the possible problem?Thanks and best regards,PS.2006/10/29, Alberto Pastore [EMAIL PROTECTED]: Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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 I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.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___--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 -- Com os melhores cumprimentos,Marco Mouta ___ --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 Manager
Yes, I did check it. The asterisk.st1 is running at /var/run folder, I tried to change to /var/run/asterisk/asterisk.ct1, then I will have the error "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?" I thing it have something with the permission. How can I check what user running under the php? How can I specify the user for PHP? What user should I specify? My /etc/asterisk/manager.conf has [admin]secret = passworddeny=0.0.0.0/0.0.0.0permit=127.0.0.1/255.255.255.0read = system,call,log,verbose,command,agent,userwrite = system,call,log,verbose,command,agent,user - Original Message - From: Lacy Moore - Aspendora To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, October 25, 2006 11:34 AM Subject: Re: [asterisk-users] Asterisk Manager Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My manager.conf has an entry. Check to make sure the file is actually /var/run/asterisk.ctl and not /var/run/asterisk/asterisk.ctl. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Diva server 4bri and Portuguese BRI
Marco Mouta ha scritto: pls post your misdn.conf as well as extensions.conf May be i can help. Sou Português:) On 10/29/06, *Pedro Silva* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Thanks Alberto! I tested with telsampl like you said (with various configurations for de diva) and this not works...:( The trace is: Enter destination address: 273xx --Conn_Req(273xx) Connect_Con-- [29]:Disc_Ind-- --Disc_Res **Call cleared*** Any idea for the possible problem? Thanks and best regards, PS. I think Pedro is not using mISDN, he's using chan_capi, but his problem arises before asterisk is involved. Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.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] VoIP GSM Gateways
Hi, On Sun, 2006-10-29 at 13:46 +0200, Tzafrir Cohen wrote: Is vISDN (extra kernel modules, extra non-standard Asterisk channel) required? The page on vGSM there suggests it is. no, vgsm uses only a part of visdn (timer system and streamport), so you need only chan_vgsm, visdn_streamport (for audio) and visdn_timer_system for timing. Other visdn things, like chan_visdn, complex visdn pci conf etc etc is not needed. Nothing more. The card is in production since months on various systems and is running very smooth :) Matteo. ___ --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] Diva server 4bri and Portuguese BRI
Olá Marco! :) 2006/10/29, Marco Mouta [EMAIL PROTECTED]: pls post your misdn.conf as well as extensions.conf The asterisk version that i used with trixbox dont't have misdn.conf... I used capi.conf. For now, i dont care about asterisk, because with the divas utility telsampl i know that the problem is between diva card and BRI access. So i need to solve first this problem and only after that im care with asterisk... :) Obrigado desde já pela disponibilidade de ajuda! PS. May be i can help. Sou Português:) On 10/29/06, Pedro Silva [EMAIL PROTECTED] wrote: Thanks Alberto! I tested with telsampl like you said (with various configurations for de diva) and this not works...:( The trace is: Enter destination address: 273xx --Conn_Req(273xx) Connect_Con-- [29]:Disc_Ind-- --Disc_Res **Call cleared*** Any idea for the possible problem? Thanks and best regards, PS. 2006/10/29, Alberto Pastore [EMAIL PROTECTED]: Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: call failed: declined. Anyone have experience with this setup? What are the main parameters for bri card configuration? D-channel protocol: ETSI-DSS1 or other? Interface mode: NT or TE? Direct Inward Dialing (DID): Yes ou no? (MSN ou DID?) Thanks by any kind of help! Best regards, PS. ___ --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 I'm not sure about Portuguese operators standard, but I bet ETSI-DSS1 should work just fine. The interface mode is surely TE. The DID/MSN should not affect outgoing calls, I generally leave DID off unless the telco company has that service active. If you're using the diva server for linux package from eicon (divas4linux, currently rel. 8.2), you should find a very simple utility named telsampl under /usr/lib/eicon/divas which you can run besides asterisk, to test outgoing calls. You should run it with this command line: telsampl -c x where x is the bri port you wish to test (1..4) then at the prompt type c and enter a pstn number, e.g. your mobile phone, then you can watch the log onscreen. If the outgoing call works, then your isdn setup is correct, and the problem is in asterisk. The message from xlite is not meaningful, as it could occur on many situations. You should watch the debug output on asterisk console. That helped me a lot. Alberto. -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.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 ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Diva server 4bri and Portuguese BRI
Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --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] H.263 Video Messages
Hello,I'm trying to set the Asterisk to leave a video message to the mailbox, but there is some compatibility problem, although h263 is identified as the matching codec, as you can see in the debug messages below:Capabilities: us - 0x80100 (g729|h263), peer - audio=0x43f (g723|gsm|ulaw|alaw|g726|adpcm|ilbc)/video=0xc (h261|h263), combined - 0x8 (h263)Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)..Oct 29 16:20:58 WARNING[3098]: channel.c:506 ast_best_codec: Don't know any of 0x8 formatsDoes anyone know how to resolve this problem?Thanks,David___ --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] hardware requirements..
Hello, I have Asterisk running on Debian Sarge. I use AGI to billing and now I´m planing to separate mysql from Asterisk Box, resulting: (Asterisk + AGI) - (Iptables + showrewall + Apache + PHP + MySQL) What hardware configuration do you recommend for the second box, in order to maintain the quality in calls? I´m planning to run upon 30 (E1) concurrent calls. Thanks in advance. R.R. Libera ___ --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] Diva server 4bri and Portuguese BRI
Finally this works!!! :) Tanks to Alberto and Marco by your help! The problems are: - the cable was connected to the wong card port... :( - the card config needs to be: ETSI; TE; Point-to-Point (I thought that was point-to-multipoint). Best regards, PS. 2006/10/29, Pedro Silva [EMAIL PROTECTED]: Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:074 - alloc cr in use =4 4:1736:076 - free cr in use =3 4:1736:077 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:078 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:078 - alloc cr in use =4 4:1736:080 - free cr in use =3 4:1736:080 - DELETEID ok: deleted Id:6 freeIds=eb 4:1736:081 - CREATEID ok: context:1f assigned Id:6 freeIds=eb 4:1736:081 - alloc cr in use =4 4:1736:083 - [1,0] dsp_assign 0016, 0, 2048 4:1736:083 - [1,0] CAI[00] 16 00 00 00 00 08 4:1736:084 - [1,0] Download 532 requested 4:1736:084 - MORE 4:1736:085 - SIG-X(029) 08 01 36 05 A1 04 03 80 90 A3 18 01 83 1E 02 80 83 70 0A 80 39 36 33 30 34 35 37 32 33 Q.931 CR36 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 83 Progress Indicator 80 83 Called Party Number 80 '963045723' 4:1736:085 - SIG-S 0-1 e:885 4:1736:087 - ACTIVATION_REQ 4:1744:147 - L1_DOWN 4:1744:150 - SIG-EVENT 08 4:1744:150 - SIG-EVENT 08 4:1744:150 - EVENT: Call failed in State 'Call initiated' Link disconnected, Layer-1 error (cable or NT) 4:1744:150 - SIG-S 1-0 e: 4:1744:151 - [1,0] dsp_release 4:1744:155 - free cr in use =3 4:1744:156 - DELETEID ok: deleted Id:6 freeIds=eb I disconnect the rj45 cable from alcatel pbx and connect that to the diva card (with alcatel pbx i can make calls normally). The green led of the diva card is activated when i connect the cable. So i dont understand why the message Link disconnected, Layer-1 error (cable or NT)... This debug is th same if the cable is connected to the NT or not. Any ideas...? Thanks! PS. ___ --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] No zap* commands?
I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? ___ --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] No zap* commands?
On 29 Oct 2006, at 20:24, Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? zap modules not loaded? try: load chan_zap.so on the console and/or put that into modules.conf ___ --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] Something is trashing /var/run
For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed zaptel, wanpipe and asterisk on a fresh install of CentOS. I did the same thing on Friday and had the same problem, so I scrubbed the disk and tried again. Any ideas what's going on here ? Thanks, Jim. ___ --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] Out bound calls 'you must first dial a 1'
Hello,
I have asterisk 1.2.9 running on a Debian sarge server, my outbound dial
plan looks something like this:
[outbound-longdistance]
exten = _91NXXNXX,1,Dial(${OUTBOUND1}/${EXTEN:1})
About every other outbound call we make, we get the 'you must first dial a
1' message from our phone provider. It only seems to happen every other try
if we try to make multiple out bound calls without waiting 5 minutes in
between.
My outbound line is set to: OUTBOUND1=Zap/2-1
Has anyone seen this or know of a fix for it? Please let me know. thanks.
George
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Re: [asterisk-users] Something is trashing /var/run
mkdir /var/run/asterisk in /etc/asterisk/asterisk.conf change where you see /var/run with /var/run/asterisk Jim Lynch wrote: For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed zaptel, wanpipe and asterisk on a fresh install of CentOS. I did the same thing on Friday and had the same problem, so I scrubbed the disk and tried again. Any ideas what's going on here ? Thanks, Jim. ___ --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] Linksys PAP2: calling tone stops after 5 tones
Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose ___ --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] No zap* commands?
Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? only a hint have you restarted asterisk? or either, have you checked that all of zaptel co has installed successfully? hope you find soon the solution, regards, Barbara F. ___ --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] No zap* commands?
Conrad Wood wrote: On 29 Oct 2006, at 20:24, Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? zap modules not loaded? try: load chan_zap.so on the console and/or put that into modules.conf Said not found. So I checked, sure enough, it isn't anywhere to be found. So I looked in the zaptel source. Nothing there so I looked in the asterisk source. Ah! a chan_zap.c but no chan_zap.so. It doesn't seem to be compiled for some reason. There don't appear to be any comments in the README or INSTALL files about configuration parameters anywhere that have to be changed. So I trashed 1.2.12 asterisk and downloaded 1.2.13. And zaptel 1.2.10. I'll see what that does. Thanks, Jim. ___ --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] Something is trashing /var/run
Rodrigo Gonzalez wrote: mkdir /var/run/asterisk in /etc/asterisk/asterisk.conf change where you see /var/run with /var/run/asterisk Jim Lynch wrote: For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed zaptel, wanpipe and asterisk on a fresh install of CentOS. I did the same thing on Friday and had the same problem, so I scrubbed the disk and tried again. Any ideas what's going on here ? Thanks, Jim. Ah! Thanks. That'll sure do it. Hope someone has filed a bug for that oversight. Jim. ___ --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] Out bound calls 'you must first dial a 1'
George Patterson wrote:
Hello,
I have asterisk 1.2.9 running on a Debian sarge server, my outbound
dial plan looks something like this:
[outbound-longdistance]
exten = _91NXXNXX,1,Dial(${OUTBOUND1}/${EXTEN:1})
This has been covered quite a bit in the archives. Some providers want
a second or two wait before the dial. Asterisk is dialing too fast.
Add a wait or two:
From the wiki:
http://www.voip-info.org/wiki/index.php?page=Asterisk+ZAP+channels
Doug
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Re: [asterisk-users] No zap* commands?
On Sun, Oct 29, 2006 at 04:07:21PM -0500, Jim Lynch wrote: Conrad Wood wrote: On 29 Oct 2006, at 20:24, Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? zap modules not loaded? try: load chan_zap.so on the console and/or put that into modules.conf Said not found. So I checked, sure enough, it isn't anywhere to be found. So I looked in the zaptel source. Nothing there so I looked in the asterisk source. Ah! a chan_zap.c but no chan_zap.so. It doesn't seem to be compiled for some reason. There don't appear to be any comments in the README or INSTALL files about configuration parameters anywhere that have to be changed. Check channels/Makefile for the tests that determine if chan_zap.so should be built. (applies for Asterisk = 1.2.x ) So I trashed 1.2.12 asterisk and downloaded 1.2.13. And zaptel 1.2.10. I'll see what that does. I'd be surprised if that is a big difference. -- 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] Asterisk Voicemail with ODBC Realtime Access
Hi
I was trying to have realtime voicemail working with ODBC Driver.Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page )
But I do not manage to make it work with ODBC.Outside Asterisk, ODBC works fine, I can access my databases tables !
Asterisk fails to start if I use pre-connect = yes in res_odbc.conf ( See errors below )If I do not use it, then, I get another error message : res_config_odbc.c: SQL Alloc Handle failed! when I try to access the voicemail.
Any ideas would be more than welcome !
Thanks !
Here is my config
/etc/odbcinst.ini[MySQL]Description = ODBC for MySQLDriver = /usr/lib/libmyodbc.soSetup = /usr/lib/libodbcmyS.soFileUsage = 1
/etc/odbc.ini[MySQLast]Description = MySQL ODBC Driver TestingDriver = MySQL#Socket = /var/run/mysqld/mysqld.sockTrace = YesTraceFile = odbc_mysql.logServer = localhost
User = asteriskuserPassword = amp109Database = asteriskrealtime#Option = 3Port = 3306
isql -v MySQLast and then help shows me my Tables correctly,SELECT queries work perfectly
/etc/asterisk/res_odbc.conf[asterisk]enabled = yesdsn = MySQLastusername = asteriskuserpassword = amp109;pre-connect = yes
/etc/asterisk/extconfig.conf[settings]voicemail = odbc,asterisk,voicemail_users
Modules in Asterisk :asterisk*CLI show modules like odbcModule Description Use Countres_config_odbc.so ODBC Configuration 1
res_odbc.so ODBC Resource 0cdr_odbc.so ODBC CDR Backend 03 modules loaded
Error Messages:
if pre-connect = yes is used in res_odbc.confOct 29 16:23:21 VERBOSE[15016] logger.c: MySQL RealTime driver loaded.Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_adsi.so]Oct 29 16:23:21 VERBOSE[15016]
logger.c: [res_adsi.so] = (ADSI Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so]Oct 29 16:23:21 VERBOSE[15016] logger.c: [res_odbc.so] = (ODBC Resource)Oct 29 16:23:21 VERBOSE[15016] logger.c
: == Parsing '/etc/asterisk/res_odbc.conf': Oct 29 16:23:21 VERBOSE[15016] logger.c: == Parsing '/etc/asterisk/res_odbc.conf': FoundOct 29 16:23:21 NOTICE[15016] res_odbc.c: registered database handle 'asterisk' dsn-[MySQLast]
Oct 29 16:23:21 NOTICE[15016] res_odbc.c: Connecting asterisk
And this is what I get on the standard error output:*** glibc detected *** malloc(): memory corruption: 0x09f4daf0 ***/usr/sbin/safe_asterisk: line 50: 14920 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.*** glibc detected *** malloc(): memory corruption: 0x0a023af0 ***/usr/sbin/safe_asterisk: line 50: 15016 Aborted (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 134Asterisk exited on signal 6.Automatically restarting Asterisk.
If no pre-connect, then I can start asterisk, but I get this error when I try to access VM:Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing VoiceMail(IAX2/telegrupp-4,
[EMAIL PROTECTED]|) in new stackOct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'
Oct 29 16:24:43 DEBUG[15178] channel.c: Avoiding initial deadlock for 'IAX2/telegrupp-4'Oct 29 16:24:43 WARNING[15262] res_config_odbc.c: SQL Alloc Handle failed!Oct 29 16:24:43 WARNING[15262] app_voicemail.c: No entry in voicemail config file for '200'
Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Goto(IAX2/telegrupp-4, exit-FAILED|1) in new stackOct 29 16:24:43 VERBOSE[15262] logger.c: -- Goto (macro-vm,exit-FAILED,1)
Oct 29 16:24:43 VERBOSE[15262] logger.c: -- Executing Playback(IAX2/telegrupp-4, im-sorryan-error-has-occured) in
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[asterisk-users] AEL2 and the variables
Hi,
I am using Asterisk 1.2.12.1 + the AEL2 patch.
If I use a variable instead of the extension itself, an incoming call cannot
be connected.
${ID-FST1} = Dial(SIP/gs|15|r); == NON ok
sip debug shows :
Looking for 6674262730 in interne (domain 192.168.1.14)
SIP/2.0 404 Not Found
Is it a bug or am I doing something wrong?
Thank you.
//===
// extensions.ael2
globals {
ID-FST1=6674262730;
GS=SIP/gs;
}
context entrant {
//6674262730 = Dial(SIP/gs,15,r); == OK
${ID-FST1} = Dial(SIP/gs|15|r); == NON ok
}
context interne {
includes {
entrant;
}
}
-
Dominique Dartois
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Re: [asterisk-users] SIP v IAX2
I had a slide on this at my astricon presentation last week (about developing our IAX/java softphone) The slide basically said that I did IAX instead of SIP because the IAX draft RFC is ~100 pages whereas there are 100 RFCs on SIP/STUN/RTP - That's a lot more reading ! (Plus of course the other IAX advantages everyone else mentioned...) Tim Panton www.mexuar.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
[asterisk-users] blind transfers with IP Polycom 501
Im running Asterisk 1.2.8 with Polycom ip501s xten softphones The only problem Im experiencing is the following: I cant seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer = # This works for all my softphones, just not the 501 phones. Soft key Blind Transfers: Then I tried blind transfers through the phone like this: Transfer Blind Key Extension. Heres the problem. If I enter a 2 digit extension, it works. Example: Transfer Blind Key 70 (this parks my calls without a problem) If I try to blind transfer an extension with three or more digits, the phone cancels the blind transfer. Is there something obvious Im missing here?? Thanks in advance. ___ --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] blind transfers with IP Polycom 501
For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote: I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer = # This works for all my softphones, just not the 501 phones. Soft key Blind Transfers: Then I tried blind transfers through the phone like this: Transfer à Blind Key à Extension. Here's the problem. If I enter a 2 digit extension, it works. Example: Transfer à Blind Key à 70 (this parks my calls without a problem) If I try to blind transfer an extension with three or more digits, the phone cancels the blind transfer. Is there something obvious I'm missing here?? Thanks in advance. ___ --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] blind transfers with IP Polycom 501
Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote: I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer = # This works for all my softphones, just not the 501 phones. Soft key Blind Transfers: Then I tried blind transfers through the phone like this: Transfer à Blind Key à Extension. Here's the problem. If I enter a 2 digit extension, it works. Example: Transfer à Blind Key à 70 (this parks my calls without a problem) If I try to blind transfer an extension with three or more digits, the phone cancels the blind transfer. Is there something obvious I'm missing here?? Thanks in advance. ___ --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] Advice on GUI
Without providing a link to the list, or citing your front-runners, you can't really expect people to reply, can you?On 10/27/06, Frédéric Blaise [EMAIL PROTECTED] wrote:Hello allI would like to know your opinions on free GUI used to manage Asterisk. Which is better?My setup is quite small, about 15-20 phones. I've seen the liste onvoip-info.Thanks all.fred___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] blind transfers with IP Polycom 501
It was the digit map. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo Romero Sent: Sunday, October 29, 2006 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] blind transfers with IP Polycom 501 Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote: I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer = # This works for all my softphones, just not the 501 phones. Soft key Blind Transfers: Then I tried blind transfers through the phone like this: Transfer à Blind Key à Extension. Here's the problem. If I enter a 2 digit extension, it works. Example: Transfer à Blind Key à 70 (this parks my calls without a problem) If I try to blind transfer an extension with three or more digits, the phone cancels the blind transfer. Is there something obvious I'm missing here?? Thanks in advance. ___ --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] blind transfers with IP Polycom 501
Yes, digitmap... If you just want to allow any digit pattern, use this digitmap: xx.T x - Any valid digit . - 0 or more ocurences of previous charracter T - Default timeout (3 seconds) Any digit followed by a 3 second timeout will match. You can include pattern to match * and #. xx.T|*x.T|#x.T Julian. On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote: Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote: I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer = # This works for all my softphones, just not the 501 phones. Soft key Blind Transfers: Then I tried blind transfers through the phone like this: Transfer à Blind Key à Extension. Here's the problem. If I enter a 2 digit extension, it works. Example: Transfer à Blind Key à 70 (this parks my calls without a problem) If I try to blind transfer an extension with three or more digits, the phone cancels the blind transfer. Is there something obvious I'm missing here?? Thanks in advance. ___ --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] Polycom IP500 Problems
Hi, Im having problems getting a polycom IP500 phone to register. Ive edited the xml files and appropriate .cfg files and it shows up registered with no ip address and if I go into the phones status menu and pick lines it says its not registered. If I tell it to upload the log file it doesn't say anything about why its not registering / what errors its getting. Any advice is appreciated. Thanks --Don Wisdom ___ --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] tx_fax not getting entire fax
Hi Jerry,I have never managed to get rx/tx fax working reliably. IAXModem (which in effect grafts steves spandsp library to hylafax) however does work well.-bradley On 10/29/06, Jerry Geis [EMAIL PROTECTED] wrote: Steve,I am trying to get tx_fax to work. I am using a TDM2401E card.I have a 3 page fax and I only receive the first page on every attempt.I think I have enabled debug output below.Can you tell me what the problem might be? I am using snapshot from oct 26. asterisk 1.2.13 and libtiff 3.6.1-12from redat/centos 4.4.THanks,Jerry-Oct 28 13:13:40 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 4Oct 28 13:13:40 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Settx type 0Oct 28 13:13:42 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Setrx type 0Oct 28 13:13:42 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 8Oct 28 13:13:43 DEBUG[22746]: chan_sip.c:1412 __sip_ack: Stoppingretransmission on '[EMAIL PROTECTED]' of Request 102: Match FoundOct 28 13:13:43 DEBUG[22738]: pbx_spool.c:319 scan_service: Delayingretry since we're currently running '??x(?'Oct 28 13:13:43 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set rx type 0Oct 28 13:13:43 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Settx type 4Oct 28 13:13:44 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Setrx type 4Oct 28 13:13:44 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Set tx type 0Oct 28 13:13:46 DEBUG[22738]: pbx_spool.c:319 scan_service: Delayingretry since we're currently running '??x(?'Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Setrx type 0 Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Settx type 1Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Setrx type 13Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX FAX exchange completeOct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX Settx type 13Oct 28 13:13:46 DEBUG[22763]: app_txfax.c:69 span_message: FLOW FAX FAXexchange completeOct 28 13:13:48 DEBUG[22745]: chan_iax2.c:7074 socket_read: Immediately destroying 3, having received hangupOct 28 13:13:48 DEBUG[22763]: app_txfax.c:256 txfax_exec: Got hangupOct 28 13:13:48 DEBUG[22763]: pbx.c:2341 __ast_pbx_run: Extensionsmvoice_faxout, priority 1 returned normally even though call was hung up ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Pager Voicemail Message
Yes. It should be in that same file. Poke around. - Original Message - From: David To: asterisk-users@lists.digium.com Sent: Sunday, October 29, 2006 12:43 PM Subject: [asterisk-users] Pager Voicemail Message Hello,In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system.Is there a way to manipulate this message, as well?Thanks,David ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Re: [OT] wi-fi ip phone scenario
WIP 330 and it sux :( - Original Message - From: Alberto Pastore [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, October 29, 2006 11:35 AM Subject: [asterisk-users] Re: [OT] wi-fi ip phone scenario Martin Joseph wrote: I think it's cleary true that wiring WIFI infrastructure is easier and more reliable then WDS. On the other hand, I have been running my little network with WDS for over three weeks now, and it has been completely reliable. The tricks where to configure things properly and to have the bases closer together then one would think would be needed. Once this was setup. It works, and it keeps working. We had a couple of stress tests also, one black out and one unplugged router (carpenter). Came up cleanly and continued working fine. No mis-registrations and no problems. Marty Can I ask you guys which phones are you using? Alberto. ___ --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] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'
Hi, I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.2.13]# so I have ' make clean' -- Then I downloaded the 'asterisk-1.4.0-beta3' --- untar it then run 'make' I get: [EMAIL PROTECTED] asterisk-1.4.0-beta3]# make make install The configure script must be executed before running 'make'. make: *** [makeopts] Error 1 [EMAIL PROTECTED] asterisk-1.4.0-beta3]# make The configure script must be executed before running 'make'. make: *** [makeopts] Error 1 Any help please? T ___ --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] blind transfers with IP Polycom 501
Thanks so mAre there any drawbacks to this? This is the digitmap I ended up using and it seems to work: [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[1-9]xxxT -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Sunday, October 29, 2006 6:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 Yes, digitmap... If you just want to allow any digit pattern, use this digitmap: xx.T x - Any valid digit . - 0 or more ocurences of previous charracter T - Default timeout (3 seconds) Any digit followed by a 3 second timeout will match. You can include pattern to match * and #. xx.T|*x.T|#x.T Julian. On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote: Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero [EMAIL PROTECTED] wrote: I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer = # This works for all my softphones, just not the 501 phones. Soft key Blind Transfers: Then I tried blind transfers through the phone like this: Transfer à Blind Key à Extension. Here's the problem. If I enter a 2 digit extension, it works. Example: Transfer à Blind Key à 70 (this parks my calls without a problem) If I try to blind transfer an extension with three or more digits, the phone cancels the blind transfer. Is there something obvious I'm missing here?? Thanks in advance. ___ --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 ___ --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-1.2.13 fails to 'Make' in Fedore Core 6'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Keithy escribió: I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.2.13]# so I have ' make clean' Have you run ./configure prior to any 'make' command? Btw, check that you have kernel headers sources developer packages from the required software (in case FC has separate packages for them); here you can have an idea (focused on Debian, though) of what is needed: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian. hope this helps, regards, BarbaraF. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFRVTVGkaV37yLzccRApmCAJ9LQScuxR0H68JtIMItRkID64T+QACaA3aA mb4SLinlZrYKQOX+7Hp4u3w= =Y6SQ -END PGP SIGNATURE- ___ --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] Multiple dial macros at the same time
I am setting up an after-hours on-call system. Someone calls in and
requests service, and while they listen to music on hold, we dial out to
several people's cell phones and home phones. We don't know if they will
be answered by the employee, or by voicemail or a
spouse/relative/child/pet. So we play a message that says press 1 to
accept the call and ask employees to train their
spouse/relative/child/pets not to press 1.
The following extract from my dialplan shows how I have this feature set
up. This is with Asterisk 1.2.13.
[macro-screen]
exten = s,1,Set(MACRO_RESULT=CONTINUE)
exten = s,2,Read(ACCEPT|press-1-to-accept|1|skip|3|1)
exten = s,3,GotoIf($[${ACCEPT}=1]?5:4)
exten = s,4,MacroExit
exten = s,5,Set(MACRO_RESULT=)
exten = s,6,Playback(please-say-hello)
[menu]
exten = _FOLS1NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20,oM(screen))
exten = s,1,Playback(welcome)
exten = s,2,Dial(LOCAL/FOLS19195551000LOCAL/FOLS19195552000,,tm)
exten = s,3,Voicemail(u301)
exten = s,4,Hangup
In order for this to work, I needed the ability to restore MACRO_RESULT
back to an unset state. For now I just hacked the Set application so
that after removing the variable from the context, it only re-creates it
if the value provided is greater than zero length. In the future I will
probably write an UnSet application to handle this more gracefully,
unless someone knows a better way to unset a variable.
This all works fine, with one small problem that is driving me batty. I
would appreciate any insight or ideas on how to solve this. Here's the
scenario:
1. Caller dials the number and hears the welcome message,
then music on hold.
2. Simultaneous calls are made to Employee A at 555-1000
and Employee B at 555-2000 (per above).
3. Both of them answer the phone.
4. Employee A presses one and hears you will now be connected,
please say hello to the caller.
5. Employee A is bridged to the caller, says hello, and begins
working with them.
6. A few seconds later, the Employee B also presses 1. He also
hears you will now be connected.
7. Employee B fails to bridge, and is hung up on.
The problem is, these are pretty urgent calls and employees are highly
motivated to make sure they get answered. Employee B doesn't know
whether the call dropped because someone else got it, or because of a
phone system problem of some sort. He is now obligated to figure out
what's up with the call and make sure someone got it. What I want
instead is for Employee B to hear an alternate message that says
someone else got the call. This gives positive confirmation that it's
not his problem, so he can roll over and go back to sleep.
I can see two ways of doing this.
1. Write a function called BridgedChannel that takes a channel ID and
returns its bridge peer channel ID, if any. This would allow me to set a
variable __PARENTCHANNEL with the channel ID of the incoming call,
before the Dial command executes. The macro, at priority 6, can then
check BridgedChannel(${__PARENTCHANNEL}). If it has a value then the
call is already bridged and we can tell the employee not to worry.
2. Have a MySQL database with a single table with two fields, varchar
channel-ID and boolean answered. When the call starts do update table
set answered=false where channel-ID=${__PARENTCHANNEL}. When an employee
dials 1, retrieve the value of answered for __PARENTCHANNEL and also set
it to true in a single transaction. If the returned value was false,
tell them to answer the call and bridge; if the returned value was true,
tell them to go back to sleep and hang up.
Solution #1 requires me to write a whole new function, and solution #2
requires a MySQL database, which is pretty big dependency for such a
simple function.
Does anyone see a simpler way of doing this, or have any ideas for other
avenues to pursue?
Thanks in advance,
-Graham
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RE: [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'
If I remember right, FC6 doesn't necessarily install the kernel source. I had this error before when I installed on my Dell server with FC6. I had to run these commands and it got past that part: yum install kernel yum install kernel source Then reboot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of barbara figueirido Sent: Sunday, October 29, 2006 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6' -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Keithy escribió: I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.2.13]# so I have ' make clean' Have you run ./configure prior to any 'make' command? Btw, check that you have kernel headers sources developer packages from the required software (in case FC has separate packages for them); here you can have an idea (focused on Debian, though) of what is needed: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian. hope this helps, regards, BarbaraF. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFRVTVGkaV37yLzccRApmCAJ9LQScuxR0H68JtIMItRkID64T+QACaA3aA mb4SLinlZrYKQOX+7Hp4u3w= =Y6SQ -END PGP SIGNATURE- ___ --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] meet me
Does it say invalid conference number, or invalid password? You need to have either digium hardware installed, or ztdummy installed. Otherwise conferencing will not work. And also it is better to install trixbox instead of AAH. ___ --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] Trixbox installation - ZAP channels becoming upresponsive
One of the reasons, which I've experienced, is that the computer's processor gets over loaded. If its not agenuine server, then this will happen. A standardcomputer made for general home or office use will give trouble on zap channels when used with any hardware which does not have its own microprocessor and depends on the computer's microprocessor. This includes digium cards as well which rely on the processing power of the computer's intel or AMD processors, hence making zap channels sometimes irresponsive for some seconds. ___ --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] Pager Voicemail Message
I looked. There's nothing there.I even did a search under /etc/asterisk for files containing "Asterisk PBX" and "New VM" (both part of the Pager message) and it didn't help. I suspect that it may be in the code.- Original Message From: Dovid B [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, October 30, 2006 2:15:03 AMSubject: Re: [asterisk-users] Pager Voicemail Message Yes. It should be in that same file. Poke around. - Original Message - From: David To: asterisk-users@lists.digium.com Sent: Sunday, October 29, 2006 12:43 PM Subject: [asterisk-users] Pager Voicemail Message Hello,In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system.Is there a way to manipulate this message, as well?Thanks,David___ --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] No ring tone when using IAX
When I call from a softphone using IAX2 there is no ring tone This is the same if I call in to the IVR and press # and dial the stations ext number no ring tone And I get the same if I call in using a DID on an IAX2 trunk BUT if I use anything that is SIP I get the ring tone Softphone DISA Trunks Let me know what I should check. -- Best regards, Al Bochter Bochter Services (Voip PBX) Free World DialUp: 780217 EXT: 250 WebSite: http://www.freeworlddialup.com/ http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] Maximum talktime in a queue?
On 10/26/06, Lenz [EMAIL PROTECTED] wrote:
When you log in a callback agent, you enter first the agent code, and then
the extension he's sitting at. The context is usually specified in the
dialplan command, but the result is that asterisk knows that agent 103 is
sitting at [EMAIL PROTECTED] [EMAIL PROTECTED] is a working extension in a
working
context, where you do something on the lines of:
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
In this dial command you're free to add whatever option you may like,
including the ones to limit call length.
I have
[sip]
exten = 605,1,Dial(SIP/605||S(30))
in my sip context (where agents are located) When I call this from
another sip phone, it works ie call gets terminated in 30 seconds, but
when this extension is called from queue it does not work.
The asterisk cli gives the following output
-- outgoing agentcall, to agent '605', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/605
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/605||S(30)) in new stack
-- Setting call duration limit to 30 seconds.
-- Called 605
-- SIP/605-082f56c0 is ringing
-- Agent/605 is ringing
As you can see the call duration is properly set in the logs.
The queue.conf has the following entries for the queue
musiconhold = default
strategy = roundrobin
servicelevel = 60
timeout = 60
retry = 5
wrapuptime=5
announce-frequency = 90
announce-holdtime = yes
member = Agent/605
Any help to get this working will be much appreciated.
raj
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[asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone
For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general] country=au ___ --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] CID and CDR conflict?
Hi all, I've been beating my head against this for some time now. For incoming calls, I'd like to send my users a localized caller id number. By localized, I mean one with out the 1+areacode for local calls and only 10 digits (minus the leading 1) for long distance calls. For example: I get a call from 1501234. Since I live in the 505, I should see: 5551234 on my caller id. However, if I get a call from 18035556789, I should see: 8035556789 on my caller id. The problem is that I also want to preserve the original calling number in my CDR(src) field. But everytime I change the CID number, it changes the CDR(src) field. Is there any way I can have the best of both worlds? TIA, Mike Diehl. ___ --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] Incorrect Ring tone. Getting a US tone when it should be AU tone
On Mon, Oct 30, 2006 at 04:13:12PM +1100, Klaverstyn, David C wrote: For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au Well, this only loads the tonezone into Zaptel. But you don't use any zaptel channels zapata.conf [channels] language=au This sets dialplan language for Zaptel channels. However it is not applied, s you don't have any channels. indications.conf [general] country=au But did you actually include a definition of that country? -- 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] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'
On Sun, Oct 29, 2006 at 07:53:21PM -0600, Ward, Bill wrote: If I remember right, FC6 doesn't necessarily install the kernel source. Actually, the kernel headers are never required to build asterisk . /usr/include/linux is not from the kernel source. I had this error before when I installed on my Dell server with FC6. I had to run these commands and it got past that part: yum install kernel yum install kernel source Huh? Here's a shorter version: yum install kernel-devel But then-again, the relevant package is glibc-kernel or something. Then reboot. No reboot is required, as you didn't install a new kernel. Oh, and from what I see, building on Ubuntu Edgy will get you the same error. -- 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] dialing external number within meetme
Use the manager API and send a call from the exten of the meetme room to the number you want to dial using the Local/ channel. MATT--- On 10/28/06, Bartosz Wegrzyn - maillists [EMAIL PROTECTED] wrote: hello, is it possible to dial out external number within running conference, for example dial out using zap channel and connect to pstn conference, thx bart ___ --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] CID and CDR conflict?
Mike Diehl wrote: Hi all, I've been beating my head against this for some time now. For incoming calls, I'd like to send my users a localized caller id number. By localized, I mean one with out the 1+areacode for local calls and only 10 digits (minus the leading 1) for long distance calls. For example: I get a call from 1501234. Since I live in the 505, I should see: 5551234 on my caller id. However, if I get a call from 18035556789, I should see: 8035556789 on my caller id. The problem is that I also want to preserve the original calling number in my CDR(src) field. But everytime I change the CID number, it changes the CDR(src) field. Is there any way I can have the best of both worlds? More than 1 ways to skin the rabbit: a. Set CALLERID(num) to localized number for and store the original cid in the CDR(userfield) b. Set CALLERID(name) to localized number. This method depends on the phone. Some phone will display only the name part if it's supplied (what you want), others will show both. The CDR(src) is preserved but CDR(clid) will look like 5551234 1505551234 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
Re: [asterisk-users] Pager Voicemail Message
On 10/29/06, David [EMAIL PROTECTED] wrote: I looked. There's nothing there.I even did a search under /etc/asterisk for files containing Asterisk PBX and New VM (both part of the Pager message) and it didn't help. I suspect that it may be in the code. I would suggest looking again. If it really isn't there, then you areprobably not using the default configuration files. That's ok. Either go to the location of your source files, or download the source. Go to the configs directory. Look in voicemail.conf and copy the needed sections out. If the needed sections are commented out or don't exist (which is probably what it is in your case), it goes by defaults. In my file, it is around line 90. Search for pagersubject. You can change the pager subject, from string, and body. ___ --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] Incorrect Ring tone. Getting a US tone when it should be AU tone
By default, the file contained the following. I'm guessing this is what you mean. [au] ringcadence = 400,200,400,2000 dial = 413+438 busy = 425/375,0/375 ring = 413+438/400,0/200,413+438/400,0/2000 congestion = 425/375,0/375,420/375,0/375 callwaiting = 425/200,0/200,425/200,0/4400 dialrecall = 413+438 record = !425/1000,!0/15000,425/360,0/15000 info = 425/2500,0/500 std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100 facility = 425 stutter = 413+438/100,0/40 ringmobile = 400+450/400,0/200,400+450/400,0/2000 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, 30 October 2006 3:26 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Incorrect Ring tone. Getting a US tone when itshould be AU tone On Mon, Oct 30, 2006 at 04:13:12PM +1100, Klaverstyn, David C wrote: For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au Well, this only loads the tonezone into Zaptel. But you don't use any zaptel channels zapata.conf [channels] language=au This sets dialplan language for Zaptel channels. However it is not applied, s you don't have any channels. indications.conf [general] country=au But did you actually include a definition of that country? -- 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
Re: [asterisk-users] Pager Voicemail Message
Hi Lacy,Thanks a lot.I may have deleted these lines from the voicemail.conf file in the past. I followed your suggestion and found these lines in the sample configuration file.Again, thank you very much.David- Original Message From: Lacy Moore - Aspendora [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comSent: Monday, October 30, 2006 8:06:22 AMSubject: Re: [asterisk-users] Pager Voicemail Message On 10/29/06, David [EMAIL PROTECTED] wrote:I looked. There's nothing there.I even did a search under /etc/asterisk for files containing "Asterisk PBX" and "New VM" (both part of the Pager message) and it didn't help. I suspect that it may be in the code. I would suggest looking again. If it really isn't there, then you areprobably not using the default configuration files. That's ok. Either go to the location of your source files, or download the source. Go to the configs directory. Look in voicemail.conf and copy the needed sections out. If the needed sections are commented out or don't exist (which is probably what it is in your case), it goes by defaults. In my file, it is around line 90. Search for "pagersubject". You can change the pager subject, from string, and body.___ --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] No ring tone when using IAX
On 22:34, Sun 29 Oct 06, Al Bochter wrote:
When I call from a softphone using IAX2 there is no ring tone
This is the same if I call in to the IVR and press # and dial the
stations ext number no ring tone
And I get the same if I call in using a DID on an IAX2 trunk
BUT
if I use anything that is SIP I get the ring tone
Softphone
DISA
Trunks
Let me know what I should check.
check your Dial call. You can add a r to the options. That
way it will generate ring tone while waiting for the other
side to pickup.
exten = s,n,Dial(IAX2/sometrunk/${NR_TO_DIAL},45,r)
--
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?
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Re: [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone
What phones are you using? It could be a phone level issue. (my aastra has a setting for AU sounds..) PaulH On Mon, 2006-10-30 at 16:13 +1100, Klaverstyn, David C wrote: For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general] country=au ___ --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] [OT] wi-fi ip phone scenario
I've made some tests with Hitachi WIP3000 and 5000, works really good with roaming (without authentification). Some parts of the AP in the mesh are wired (no WDS), some others are not (using WDS), but all use the same SSID and channel. In all cases roaming was fast, quite not possible to hear it. Besides, with UTstarcom, roaming in the same mesh was not working well. Hope it helps Alban Alberto, you should have bought a dect solution, the dect technology is far better at swapping between cells. Wifi is still a little immature at this time. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Pastore Sent: Saturday, 28 October 2006 6:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [OT] wi-fi ip phone scenario Andrew Joakimsen ha scritto: Are you using WDS? While it won't totally fix every issue, I've found in my trials that turning off WDS and making sure all the AP were connected to the same wired network was way more reliable, no more random unregistartion and issue with registering (still seems to unregister at times, but re-registartion won't require a reboot). Nope. I've a group of 12 indoor colubris access points + 2 outdoor ones, and a msc5200 controller unit (the area is quite wide with two 4-storey buildings, two 8 sq.feet hangars, one 4 sq.feet outdoor parking lot) no wds, all of them are wired to ethernet switches, no wireless bridging. Diversity on all APs, automatic transimt power and channel selection (that seems to work when monitoring devices). I've tried linksys wip300/wip330, nokia e60/e70, utstarcom f1000/f1000g, samsung wip6000. The main problem (apart from firmware bugs/crashes which I hope should be fixed on newer versions) is that phones tend to stick to their associated AP even when it's clearly time to move to the next AP: if you watch the phone's rssi indicator while you walk inside the coverage area, you can see the value decreasing to almost no signal before reassociating, which is unacceptable. The only phones which seem to deal very well with it are the nokia eSeries. Unfortunately they have many other major issues. (two weeks ago I gave 3 firmware-updated e60 to my bosses to replace their cellphones and after one week they almost threw them back to me, complaining about the fact they had to reboot the phone at least twice a day because it freezed...) Also, on asterisk the qualify=2000 sip setting seems to be to low, as the console shows repeated LAGGED/UNREACHABLE/REACHABLE notices on those phones. From this experience I think wi-fi technology is not really mature enough to replace dect right now. Maybe I'm wrong. At least I hope so, my bosses are not really happy with the new system. ___ --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] Call from internal num. to VoIP gate
Greetings to All! Help to solve a problem: There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate 800 2FXS). In sip.conf they are registered so: [3301] type=friend host=172.222.135.11 username=3301 secret= defaultip=172.222.135.11 dtmfmode=rfc2833 context=it callerid=VoIPGate2Line1 3301 allow=g723.1 [3302] type=friend host=172.222.135.11 username=3302 secret= defaultip=172.222.135.11 dtmfmode=rfc2833 context=it callerid=VoIPGate2Line2 3302 allow=g723.1 [3440] type=friend host=dynamic username=3440 secret= defaultip=10.11.11.10 dtmfmode=rfc2833 context=it callerid=VoIPGateLine1 3440 allow=g723.1 ; Asterisk only supports g723.1 pass-thru! [3441] type=friend host=dynamic username=3441 secret= defaultip=10.11.11.10 dtmfmode=rfc2833 context=it callerid=VoIPGateLine2 3441 allow=g723.1 [3442] type=friend host=dynamic username=3442 secret= defaultip=10.11.11.10 dtmfmode=rfc2833 context=it callerid=VoIPGateLine3 3442 allow=g723.1 [3443] type=friend host=dynamic username=3443 secret= defaultip=10.11.11.10 dtmfmode=rfc2833 context=it callerid=VoIPGateLine4 3443 allow=g723.1 When I call from internal telephone to some of this numbers - the call goes, but when I pickup phone - communication breaks... Please, help to understand! Here a log: -- SIP read from 172.222.135.11:5060: SIP/2.0 200 OK From: Unknownsip:[EMAIL PROTECTED];tag=as2335e618 To: sip:[EMAIL PROTECTED];tag=ac16230b-13c4-ad-2c457-4966 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 10.11.0.150:5060;rport=5060;branch=z9hG4bK0e38ec58 Supported: replaces User-Agent: FXS_GW (2sipfxs.112) Contact: sip:[EMAIL PROTECTED]:0 Content-Type: application/sdp Content-Length: 220 v=0 o=FXS_GW 12367 0 IN IP4 172.222.135.11 s=Audio Session i=Audio Session c=IN IP4 172.222.135.11 t=0 0 m=audio 0 RTP/AVP 4 101 a=ptime:30 a=fmtp:101 0-11 a=rtpmap:4 G723/8000/1 a=rtpmap:101 telephone-event/8000 --- (11 headers 11 lines)--- Found RTP audio format 4 Found RTP audio format 101 Peer video RTP is at port 172.222.135.11:65535 Found description format G723 Found description format telephone-event Capabilities: us - 0x8010f (g723|gsm|ulaw|alaw|g729|h263), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing sip:[EMAIL PROTECTED]:0 for address/port to send to set_destination: set destination to 172.222.135.11, port 0 Transmitting (no NAT) to 172.222.135.11:0: ACK sip:[EMAIL PROTECTED]:0 SIP/2.0 Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK3f42da36;rport From: Unknown sip:[EMAIL PROTECTED];tag=as2335e618 To: sip:[EMAIL PROTECTED];tag=ac16230b-13c4-ad-2c457-4966 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Oct 30 08:15:34 WARNING[19154]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0x6f39d3c8 (len 387) to 172.222.135.11:0 returned -1: Invalid argument Retransmitting #4 (no NAT) to 172.222.135.11:0: BYE sip:[EMAIL PROTECTED]:0 SIP/2.0 Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK4ee204c1;rport From: Unknown sip:[EMAIL PROTECTED];tag=as2335e618 To: sip:[EMAIL PROTECTED];tag=ac16230b-13c4-ad-2c457-4966 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Oct 30 08:15:35 WARNING[19154]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0x8151630 (len 387) to 172.222.135.11:0 returned -1: Invalid argument Destroying call '[EMAIL PROTECTED]' -- SIP read from 172.222.135.11:5060: BYE sip:[EMAIL PROTECTED] SIP/2.0 From: sip:[EMAIL PROTECTED];tag=ac16230b-13c4-ad-2c457-4966 To: Unknownsip:[EMAIL PROTECTED];tag=as2335e618 Call-ID: [EMAIL PROTECTED] CSeq: 1 BYE Via: SIP/2.0/UDP 172.222.135.11:5060;branch=z9hG4bK-c0-30e94-d54 Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (2sipfxs.112) Content-Length: 0 --- (10 headers 0 lines)--- Sending to 172.222.135.11 : 5060 (non-NAT) Transmitting (no NAT) to 172.222.135.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.222.135.11:5060;branch=z9hG4bK-c0-30e94-d54;received=172.222.135.11 From: sip:[EMAIL PROTECTED];tag=ac16230b-13c4-ad-2c457-4966 To: Unknownsip:[EMAIL PROTECTED];tag=as2335e618 Call-ID: [EMAIL PROTECTED] CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Retransmitting #5 (no NAT) to 172.222.135.11:5060: BYE sip:[EMAIL PROTECTED]:0 SIP/2.0 Via: SIP/2.0/UDP 10.11.0.150:5060;branch=z9hG4bK4ee204c1;rport From: Unknown sip:[EMAIL PROTECTED];tag=as2335e618 To: sip:[EMAIL PROTECTED];tag=ac16230b-13c4-ad-2c457-4966 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP read from 172.222.135.11:5060: SIP/2.0 481 Call Leg/Transaction
[asterisk-users] Re: Linksys PAP2: calling tone stops after 5
Message: 7 Date: Sun, 29 Oct 2006 22:00:22 +0100 From: Jose Limeres [EMAIL PROTECTED] Subject: [asterisk-users] Linksys PAP2: calling tone stops after 5 tones To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose Hi Jose, i got the same expect that the tone stops after 2 tones, everything work well but the tone just stops. I´ve read something about setting the Sticky 183 to No ... check this please. In my case STicky is off - but the problem still exists :( Greets to you, Stefan ___ --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] [OT] wi-fi ip phone scenario
Alban ha scritto: I've made some tests with Hitachi WIP3000 and 5000, works really good with roaming (without authentification). Some parts of the AP in the mesh are wired (no WDS), some others are not (using WDS), but all use the same SSID and channel. In all cases roaming was fast, quite not possible to hear it. Besides, with UTstarcom, roaming in the same mesh was not working well. Hope it helps Alban So you're all using access points on the same channel? I've been told that to make a good roaming wifi lan access points must have coverage areas overlapping with the next ap for a good 20%-25% of it (just to let the client roam to another AP while moving), and they should use different channels according their topology, in order to minimize adiacent channel interference, for instance something like this 1--13-4 | | | | | | | | | 10-6--9 | | | | | | | | | 8--12-2 I'm confused... ___ --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
