[asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
On Thu, Mar 10, 2011 at 12:20 AM, Faisal Hanif fai...@vopium.com wrote: It just have ACL concept. You can add permitted IPs List to any peer then only from that IPs user can register. If you want to permit all you can add 0.0.0.0 to ACL Thanks. but could you be a little more specific? I have added the local net 172.16.16.0/24 almost everywhere I can think of, but it keeps giving that error. Even in sip.conf in the template for company IP phones, I've added contactpermit as well as just permit=172.16.16.0/24 but it still complains about that *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RR *Sent:* Thursday, March 10, 2011 7:04 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com wrote: You can add following line to your peers configuration permit=0.0.0.0/0.0.0.0 It will allow to use that peer’s account from any IP Thanks. But Like I said, that's all done. Here's the Endpoint config: [authentication] [basic-options](!); a template dtmfmode=rfc2833 context=Phones type=friend contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 deny=0.0.0.0/0.0.0.0 permit=172.16.16.0/24 host=dynamic qualify=no insecure=port,invite [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes directmedia=no [555](natted-phone) secret=$$ecret$$ disallow=all allow=ulaw allow=gsm no deal! The irony is that we have a similar configuration at another place, but we didn't need to put anything there and the phones register regardless! Is this broken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mirrors in Australia?
On Mon, Mar 7, 2011 at 2:11 AM, Stuart Longland redhat...@gentoo.orgwrote: http://mirror.aarnet.edu.au/pub/gentoo/distfiles/asterisk-1.8.3.tar.gz I haven't checked that URL, but it should be correct. That, and that mirror should be unmetered if you're on a university network. Thanks mate, it works :) Although we were in a bit of a hurry so we bit the bullet and downloaded it from downloads.asterisk.org and it had blazing speed. Downloaded at about 1.93 MB/s But will use the one you suggested in the future. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/07/2011 03:35 PM, RR wrote: Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Have you specified any '-march' or '-mcpu' options to the compiler? This sort of thing can occur if you are building for a plain-jane i386 processor or something similar. Hey Kevin, nope, nothing...just doing the standard ./configure; make menuselect; make this is a Sun Sparc v240 machine running Debian 6.0 Squeeze sparc64-smp kernel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On Mon, Mar 7, 2011 at 5:34 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/07/2011 04:31 PM, RR wrote: On Mon, Mar 7, 2011 at 5:25 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: Please do not reply directly to posters on the mailing list unless they request it. Sorry, the default on my gmail is Reply All and usually I remove relevant parties from the To/Cc: headers, guess missed it this time. Wasn't intentional. On 03/07/2011 03:35 PM, RR wrote: Hello all, mmm a bit embarrassing about not having a clue as to why we're getting this error on make of 1.8.3 [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Have you specified any '-march' or '-mcpu' options to the compiler? This sort of thing can occur if you are building for a plain-jane i386 processor or something similar. Hey Kevin, nope, nothing...just doing the standard ./configure; make menuselect; make this is a Sun Sparc v240 machine running Debian 6.0 Squeeze sparc64-smp kernel Someone with SPARC experience will have to chime in then... for some reason the configure script has determined that your compiler provides atomic instructions, but they aren't being found at link time. Ok...thanks. Is there no way for me to tell the compiler or provide flags in ./configure that can tell it to not do that? Conversely can I use -march and/or -mcpu kind of options to make this compile for my platform? If so, then what would the value be of these options or are there no values for them and one just specifies them? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
[asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
Hello all, Figured I'd repost this with an edited subject line, to attract attention of people with Debian On Sparc experience. Apologies in advance if this kind of thing is frowned upon :) [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o mpool/mpool.o recno/rec_close.o recno/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o - libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o logger.o manager.o md5.o netsock.o netsock2.o pbx.o plc.o poll.o privacy.o rtp_engine.o say.o sched.o security_events.o sha1.o slinfactory.o srv.o ssl.o stdtime/localtime.o strcompat.o strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk astobj2.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' astobj2.o:/usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: more undefined references to `__sync_fetch_and_add_4' follow utils.o: In function `ast_atomic_dec_and_test': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:635: undefined reference to `__sync_sub_and_fetch_4' utils.o: In function `ast_atomic_fetchadd_int': /usr/src/asterisk-1.8.3/include/asterisk/lock.h:589: undefined reference to `__sync_fetch_and_add_4' collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
On Mon, Mar 7, 2011 at 5:52 PM, Ernie Dunbar maill...@lightspeed.ca wrote: Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers = mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: Don't know what res_mysql.conf is, I think it should be res_config_mysql.conf? Sorry it's been a LONG time since I configured/used realtime and that also was with ODBC and TDS but I know that the file res_config_mysql.conf should definitely be there HTH \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 08:49, RR wrote: Any idea where this is coming from? seems like something is selected that doesn't have other related stuff unselected? no clue where to start looking No SPARC expert, but I seem to recall the lowest-common-denominator SPARCs lack things like hardware multiply in the instruction set. Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 09:21, RR wrote: Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org mailto:redhat...@gentoo.org wrote: Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? Normally it's specified in the environment; so maybe CFLAGS=-mcpu=v9 ./configure… I tried both ways, my way and yours i.e. setting them as env variables and it still gets that error. Also found some other stuff on the net related to that in different context but none of those work for me. Some where in some old debian archives there's some mention of the Boost libraries and the flag that must be used on Sparc with Boost libraries. Although it also says that it was fixed in some later release which was back in 2008, so am assuming that fix is still in place in Squeeze. BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Maybe, the fact that it detected 'sparc64' probably is more a case of telling the build system that the system is big-endian, requires that data structures be 64-bit aligned, etc. Use of features that weren't in the first SPARC is an optional extra. Ok, if that doesn't help then another interesting insight is that in config.log, it says that the response to 'arch' and 'arch -k' commands is 'unknown'. Don't know if that helps. Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) Telepathy; seems we think alike. :-D Must be due to me being from the same part of the world. Possibly :) although I have found that there's not a lot of activity in that list on a regular basis. So not sure if my problem will get resolved there or not :( -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
On Mon, Mar 7, 2011 at 5:45 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 03/07/2011 04:41 PM, RR wrote: Someone with SPARC experience will have to chime in then... for some reason the configure script has determined that your compiler provides atomic instructions, but they aren't being found at link time. Ok...thanks. Is there no way for me to tell the compiler or provide flags in ./configure that can tell it to not do that? Conversely can I use -march and/or -mcpu kind of options to make this compile for my platform? If so, then what would the value be of these options or are there no values for them and one just specifies them? The answer to all of those questions is probably 'yes', but that's why I said someone with SPARC experience would have to chime in. Ok, so this is solved! The culprit was the the line mcpu=v8 in the Makefile. Comment that out, and it makes properly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Solved] Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
On Mon, Mar 7, 2011 at 6:48 PM, RR ranjt...@gmail.com wrote: On Mon, Mar 7, 2011 at 6:31 PM, Stuart Longland redhat...@gentoo.orgwrote: On 03/08/11 09:21, RR wrote: Hello Stuart On Mon, Mar 7, 2011 at 6:09 PM, Stuart Longland redhat...@gentoo.org mailto:redhat...@gentoo.org wrote: Even if it doesn't help fix the problem, you probably will want to use at least -mcpu=v9 (educated guess looking at the gcc manpage) if it's an UltraSPARC as that will give you some of these instructions. Asterisk strikes me as an application that'd make fairly hefty use of things like integer multiplication. Ok, where would I put this -mcpu=v9 in the configure line? I tried ./configure CFLAGS=-mcpu=v9? Normally it's specified in the environment; so maybe CFLAGS=-mcpu=v9 ./configure… I tried both ways, my way and yours i.e. setting them as env variables and it still gets that error. Also found some other stuff on the net related to that in different context but none of those work for me. Some where in some old debian archives there's some mention of the Boost libraries and the flag that must be used on Sparc with Boost libraries. Although it also says that it was fixed in some later release which was back in 2008, so am assuming that fix is still in place in Squeeze. BTW, at the end of the configure script, it's already detecting the host cpu as sparc64. If that helps. Maybe -march needs to be specified somewhere? Maybe, the fact that it detected 'sparc64' probably is more a case of telling the build system that the system is big-endian, requires that data structures be 64-bit aligned, etc. Use of features that weren't in the first SPARC is an optional extra. Ok, if that doesn't help then another interesting insight is that in config.log, it says that the response to 'arch' and 'arch -k' commands is 'unknown'. Don't know if that helps. Another place to ask might be the Debian-SPARC mailing list? haha funny, I was just writing an email to that list when your email hit my inbox :) Telepathy; seems we think alike. :-D Must be due to me being from the same part of the world. Possibly :) although I have found that there's not a lot of activity in that list on a regular basis. So not sure if my problem will get resolved there or not :( Ok, so this is solved! The culprit was the the line mcpu=v8 in the Makefile. Comment that out, and it makes properly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mirrors in Australia?
Hello All, wondering if anyone knows of any reliable mirrors to download asterisk from in Australia or somewhere close to it than having to download stuff all the way from the US? Cheers, \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Microsoft Speech Server/UCMA Integration
Hello All, I was wondering if anyone's tried to use OR currently use the Microsoft Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines etc. If yes, then what's their experience? Please Note, this does NOT need to be integrated with Asterisk ala MRCP or some module/plugin etc. I just wanted to know if someone's used it and and what their experience has been in both, TTS and ASR. Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Tue, Jan 25, 2011 at 6:59 AM, Andrew Latham lath...@gmail.com wrote: Thanks Dave. Sounds like a man who's not had his hand soaking in ivory liquid and been through the toils and tortures of various upgrades over the years. Very insightful though. Goof thing this discussion ensued as I am learning a lot about what to be wary of not least of all, the truth about testing, RC and stable distribution. Which is why, despite eating humble pie re: the RC vs Stable discussion, I was going to wait till the status on RC changes to stable and maybe even help out a bit in the upgrade path testing. Good thing is that I don't necessarily need to muck around with the Production machines at the moment as all development is being done in the Lab, and some of that is in VMs, so I have the power of snapshots with me along with physical access to machines should anything break badly. The production machines are sitting 10,000 miles away so the best I have is console access to them. Speaking of in-place upgrades, does adding the Squeeze repo. in the sources.lst conf and running 'aptitude safe-upgrade/full-upgrade' automaticaly begins the upgrade or is there more to it? You mentioned about backing up configs and data etc so it doesn't sound like it's that simple eh? -- pretty easy... Lenny to Squeeze (5.0 to 6.0 for the mortals out there..) 1. aptitude update 2. aptitude upgrade 3. aptitude clean 4. sed -i 's/lenny/squeez/g' /etc/apt/sources.list 5. aptitude update 6. aptitude install apt dpkg aptitude 7. aptitude full-upgrade 8. aptitude clean 9. init 6 10. have a lovely beverage and relax... :) 1. A cold-stone creamery hot chocolate satchet (70 cal) 2. 2 tbps of fat free half-and half 3. 1 tbsp of instant coffee 4. 1.5 packet of splenda 5. Hot water makes an amazingly cozy low-cal beverage esp. when it's snowing outside like it is in NYC right now :) Thanks for that How-To Andrew. Appreciate it. Will have this going on, on one of the VMs with Lenny and keep up with both side by side to see if both are equally stable before I put one of them in production. Cheers, \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 3:11 AM, Stelios Koroneos skoron...@digital-opsis.com wrote: On Mon, 2011-01-24 at 01:09 -0500, RR wrote: On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.com wrote: On 11-01-23 10:24 PM, RR wrote: email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp. coz I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I see, the following If you read CHANGES, you will also see you kernel 2.6.25+ *and* glibc 2.8+. Lenny ships with 2.7-1 yep, had read that too, just very new to debian so was fearing I'll have to do a manual install / upgrade of glibcI guess that's what I have to do :( will figure out how to do that. Just an FYI. Be sure to test it to a non production system, trying to replace glibc from source is not an easy task. *MANY* things need tweaking and lots of apps can break with the wrong glibc version. Thanks for the warning Stelios. Yes, This is a VM which I snapshot every step of the way to revert back to if I break something too bad. it's a lot easier to just revert to snapshot in 20 secs, then trying to fix whatever broke :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote: On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote: In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel At this point the easiest option will be to upgrade to squeeze. R Umm yeah that might not be a smart thing to do since eventually all of this needs to run in a production environment and Squeeze is still in a RC mode. Would be nice if I could go to it though but don't think it'll be that smart esp. all other software that needs to work along with it might break too...who knows -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com wrote: On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote: On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote: In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel At this point the easiest option will be to upgrade to squeeze. R Umm yeah that might not be a smart thing to do since eventually all of this needs to run in a production environment and Squeeze is still in a RC mode. Would be nice if I could go to it though but don't think it'll be that smart esp. all other software that needs to work along with it might break too...who knows Wow, alright, after an all-nighter, I was able to get timerfd.so compiled in Asterisk 1.8.2.2 under Debian Lenny 5.0.7 with Kernel 2.6.26-2-amd64. Of course, due to the glibc requirement of 2.8+, a lot of dodgey upgrades had to be performed. I have no idea how stable this is going to be in production but I am going to write a quick How-To and stick it on the Wiki if someone can point me to the correct location this should go to. A lot of components needs to get upgraded in the correct order to have this work well, but it might save someone else the time and effort. Will respond to this email again, with the link to the Wiki page once I am done with the HowTo and people tell me where it needs to go. Cheers, \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org mailto:ro...@firedrake.org wrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a
stable/stock lenny
installation with the updated glibc as well as the latest kernel
At this point the easiest option will be to upgrade to squeeze.
R
Umm yeah that might not be a smart thing to do since eventually all
of this needs to run in a production environment and Squeeze is
still in a RC mode. Would be nice if I could go to it though but
don't think it'll be that smart esp. all other software that needs
to work along with it might break too...who knows
This a statement we hear from people periodically that just confuses me...
they say they can't update to an 'RC' release of something (Linux distro,
Asterisk, etc.) because they need to run in production mode, but they're
willing to consider replacing something as fundamental as the Linux kernel
(a bit scary) or glibc (very scary) instead.
haha touché Kevin :) Mate, the response to that is one word: Ignorance :)
people like me, who're not developers nor experts of the platform have
absolutely no clue what glibc actually does or the impact it actually has.
Nor do I know, as a user, how stable Squeeze RC2 really is at this stage of
its development. If I had more people in the community say that they're
running it in production, then maybe I'll just believe them and start
working with Squeeze directly instead of wasting my time like I did trying
to have it compiled in Lenny. I just believed when the developers of Debian
say that Squeeze RC2 is in testing and Lenny is stable and decide that
it's probably not a good idea to run RC2 in production. I guess part of the
thinking was that other software besides {*} that needs to run on this
machine may not even build or run or be stable on Squeeze RC till the
authors/users of that other software state that it's been tested with it and
it's stable or even builds on it. So, people like me believe that if I
upgrade ALL components that depend on glibc and that glibc depends on to the
current version, then we'll be ok but we wouldn't have touched anything else
in the system, not realising or understanding that satsisfying dependencies
doesn't mean anything and something somewhere could just break because of
this unsolicited upgrade thus making the system more unstable. I have really
no explanation for you as to why people (incl. myself) say these things
other than just lack of insight and knowledge about the intricacies of
things like glibc and the impact it can have on the stability of the system
when upgraded out of context. *sigh* :(
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Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 7:07 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 12:46 PM, RR wrote:
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com
mailto:ranjt...@gmail.com mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org mailto:ro...@firedrake.org
mailto:ro...@firedrake.org mailto:ro...@firedrake.org
wrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a
stable/stock lenny
installation with the updated glibc as well as the latest kernel
At this point the easiest option will be to upgrade to
squeeze.
R
Umm yeah that might not be a smart thing to do since
eventually all
of this needs to run in a production environment and Squeeze is
still in a RC mode. Would be nice if I could go to it though
but
don't think it'll be that smart esp. all other software that
needs
to work along with it might break too...who knows
This a statement we hear from people periodically that just confuses
me... they say they can't update to an 'RC' release of something
(Linux distro, Asterisk, etc.) because they need to run in
production mode, but they're willing to consider replacing something
as fundamental as the Linux kernel (a bit scary) or glibc (very
scary) instead.
haha touché Kevin :) Mate, the response to that is one word: Ignorance
:) people like me, who're not developers nor experts of the platform
have absolutely no clue what glibc actually does or the impact it
actually has. Nor do I know, as a user, how stable Squeeze RC2 really is
at this stage of its development. If I had more people in the community
say that they're running it in production, then maybe I'll just believe
them and start working with Squeeze directly instead of wasting my time
like I did trying to have it compiled in Lenny. I just believed when the
developers of Debian say that Squeeze RC2 is in testing and Lenny is
stable and decide that it's probably not a good idea to run RC2 in
production. I guess part of the thinking was that other software
besides {*} that needs to run on this machine may not even build
or run or be stable on Squeeze RC till the authors/users of that other
software state that it's been tested with it and it's stable or even
builds on it. So, people like me believe that if I upgrade ALL
components that depend on glibc and that glibc depends on to the current
version, then we'll be ok but we wouldn't have touched anything else in
the system, not realising or understanding that satsisfying dependencies
doesn't mean anything and something somewhere could just break because
of this unsolicited upgrade thus making the system more unstable. I have
really no explanation for you as to why people (incl. myself) say these
things other than just lack of insight and knowledge about the
intricacies of things like glibc and the impact it can have on the
stability of the system when upgraded out of context. *sigh* :(
And you've made my point: You chose a specific version of Debian to run,
which you are happy running in 'production'. Given that you have made that
choice, you can *only* install packages that distribution provides on your
system. Any other packages you install are not part of that version, and
thus have not gone through the same testing/qualification processes
(whatever they may be). Discussing installation of packages (any packages)
from a later Debian release, or installation of a package from source that
overwrites the Debian package, seems totally inconsistent with being 'in
production', no matter how small or large the package may be. Each such
decision must be thoroughly researched and the possible ramifications
understood before any changes are made, so as to keep the system as stable
as possible.
In essence, this is somewhat like buying a car with a high efficiency
powertrain because you want to save fuel, but then later complaining that it
doesn't accelerate as fast as you'd like... so you make plans to replace the
engine. Sure, you can do it, but you've defeated the purpose of the choice
you made in the first place :-)
I know right? I wish I could have those hours of the night back that I
wasted in trying to get it working on Lenny ... wish I'd done some homework
and realised that all sorts of Squeeze installation ISOs are in fact
available for Sparc. I thought currently only Lenny was available for Sparc
so needed to stick with it. Oh well, that's a lesson for me right there. But
hopefully not all was a wasted effort
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 8:57 PM, Dave Platt dpl...@radagast.org wrote:
I know this is an {*} list but does anyone know if simply adding the
Squeeze
repository to my sources.lst and running an 'aptitude
upgrade/safe-upgrade/full-upgrade will just upgrade Lenny - Squeeze
without me having to rebuild the system from scratch?
In my experience: you're likely to run into a few things which
need some amount of manual fiddling, after an upgrade of this sort,
but it's usually quite manageable.
The Debian people seem to be very good about making sure that
stable-version-to-stable-version upgrades go smoothly... the
process isn't perfect (from what I've seen) but it's usually
quite close. The upgrade path is usually tested out quite well
before the release team throws The Big Switch, and there normally
are good release notes which describe the corner cases which may
need manual intervention.
I have several systems which have been through multiple major
Debian upgrades, without having to be slagged down and rebuilt
from the ground up. That's better than I ever achieved with (e.g.)
Red Hat, which (in my experience) really didn't take at all well to
in-place upgrades... I usually had to do a fresh install and then
port my personal files over.
Things may not be as smooth when jumping from Stable to Testing,
precisely because this isn't an official-release pathway, and
the packages in Testing are usually in somewhat of a state of
flux. Even upgrades *within* the Testing distribution can leave
you with a system which doesn't fly right... this isn't common but
it does happen. For example, a recent upgrade within Stable pulled
a bunch of the firmware files out of the kernel package and moved
them to a separate non-free package - if I hadn't noticed an error
message during RAMdisk rebuilt, my next boot would have left me
with a non-functioning wired Ethernet adapter.
If you decide to follow this route, follow the Debian instructions
for upgrading... back up your package configurations, and (I suggest)
everything in the /etc/ directory hierarchy, as well as all of your
personal files. This will give you a much better chance to invoke
the spirit of the ancient pagan god DoOver, if something goes wrong
during the upgrade.
Thanks Dave. Sounds like a man who's not had his hand soaking in ivory
liquid and been through the toils and tortures of various upgrades over the
years. Very insightful though. Goof thing this discussion ensued as I am
learning a lot about what to be wary of not least of all, the truth about
testing, RC and stable distribution. Which is why, despite eating humble
pie re: the RC vs Stable discussion, I was going to wait till the status on
RC changes to stable and maybe even help out a bit in the upgrade path
testing. Good thing is that I don't necessarily need to muck around with the
Production machines at the moment as all development is being done in the
Lab, and some of that is in VMs, so I have the power of snapshots with me
along with physical access to machines should anything break badly. The
production machines are sitting 10,000 miles away so the best I have is
console access to them.
Speaking of in-place upgrades, does adding the Squeeze repo. in the
sources.lst conf and running 'aptitude safe-upgrade/full-upgrade'
automaticaly begins the upgrade or is there more to it? You mentioned about
backing up configs and data etc so it doesn't sound like it's that simple
eh?
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[asterisk-users] Asterisk on Debian Lenny with timerfd
Hello All, I'm sure this has been talked about and based on some searching of archives, I'd discovered that to be able to use timerfd, one needs to have a kernel version =2.6.27? Is this true? If yes, then is there anyone who's got it working in Lenny 5.0.7? Do I need to download and build the linux kernel (currently at 2.6.37) from scratch and get access to the TimerFD source? Should I even bother with it for app_confBridge or does pthread work well enough? Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Sun, Jan 23, 2011 at 10:16 PM, Paul Belanger pabelan...@digium.comwrote: On 11-01-23 10:01 PM, RR wrote: I'm sure this has been talked about and based on some searching of archives, I'd discovered that to be able to use timerfd, one needs to have a kernel version =2.6.27? Is this true? Kernel version 2.6.25 or newer, as documented in CHANGES. Thanks Paul, yes I'd read that in the CHANGES doc. But I saw some otlder email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp. coz I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I see, the following in config.log configure:27550: checking for timerfd support configure:27584: gcc -c -g -O2 conftest.c 5 conftest.c:243:25: error: sys/timerfd.h: No such file or directory conftest.c: In function 'main': conftest.c:247: error: 'NULL' undeclared (first use in this function) conftest.c:247: error: (Each undeclared identifier is reported only once conftest.c:247: error: for each function it appears in.) # uname -r 2.6.26-2-amd64 Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.comwrote: On 11-01-23 10:24 PM, RR wrote: email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp. coz I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I see, the following If you read CHANGES, you will also see you kernel 2.6.25+ *and* glibc 2.8+. Lenny ships with 2.7-1 yep, had read that too, just very new to debian so was fearing I'll have to do a manual install / upgrade of glibcI guess that's what I have to do :( will figure out how to do that. Thanks \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Debian Lenny with timerfd
On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger pabelan...@digium.comwrote: On 11-01-23 10:24 PM, RR wrote: email from Kevin Flemming talking about =2.6.27 so thought I'd ask esp. coz I have 2.6.26-2 yet I don't think I have timerfd on my machine...and I see, the following If you read CHANGES, you will also see you kernel 2.6.25+ *and* glibc 2.8+. Lenny ships with 2.7-1 In the meantime, does anyone have a nice way to update a stable/stock lenny installation with the updated glibc as well as the latest kernel (this obv. is not necessary, but couldn't hurt to have the latest :). Sorry for asking help like a bum, but I have spent an hour messing around with downloading the .deb file for the libc6 and figuring out how to install it and in turn ended up messing up my environment. Good thing I had snapshot from before, that I could restore and get back to stock. I'm a total newbie in debian so any help in some aptitude/dpkg magic to install the latest libc6 (glibc) with its dependencies on this system would be greatly appreciated :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hi Tilghman, Btw w.r.t to the patch delivered for this bug, as I stated in the notes, it worked for trunk. I tried it for 1.6.2.15 and the patch came up with a few errors, as in the patch wasn't clean and I just looked at the configure.ac.rej file and made the changes manually. I wanted to test building this on Solaris 10 u9, but wasn't able to due to my messed up dev environment. I will fix this environment and test compiling and building it assuming I made the changes that the patch was supposed to make correctly. Will let you know . I was going to add that as a note to the bug report itself but then I got distracted with something else and now it's closed and I'll have to repoen it to add any more notes. Just FYI. \RR On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.com wrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.org and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
BTW, the issue was created yesterday, but I didn't think there was a need to post it here but nevertheless for posterity, the Issue ID is: 18442 Thanks \RR On Wed, Dec 8, 2010 at 6:57 PM, RR ranjt...@gmail.com wrote: On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.comwrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.org and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org G'day Tilghman, Thanks for that thread. I guess a few other things broke because of the change and the consuming application then needs to be a little smarter like you said (and suggested by darrenr) to detect whether you're on OSOL or Solaris. Does that mean I should check this same thing out on Solaris 10 as well and see what happens? I am so lost with the Solaris build environment as (and I whinged about this earlier too) there is no good way of obtaining the standard Solaris packages and dependancies and everything just goes all over the place and then one is left scurrying around to find where the damn library needs to be for it to compile. Anyway, I will open an issue and reference this thread and we'll go from there. BTW, THANK YOU for taking note of this and trying to help. You guys will have bottomless beer pitchers paid for if you guys help me get this working and are ever in the NY area :) Cheers, \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
On Thu, Dec 9, 2010 at 10:02 AM, Bruce McAlister bruce.mcalis...@blueface.ie wrote: Hi RR, I’ve not tried compiling 1.8.1-rc1 on Solaris yet and I’ve not come across this issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build error’s though. I’m not sure if the code has changed that much between 1.8.0-rc5 and 1.8.1-rc1. I’m no coding guru by anyone’s standards, but I do build a couple applications for Solaris. What has made my life a hell-of-a-lot easier is JDS-CBE and SFE, check out the following 2 links: http://dlc.sun.com/osol/jds/downloads/cbe/ http://pkgbuild.sourceforge.net/spec-files-extra/ What the above does is setup a common build environment for building applications. The SFE (spec-file-extra) is a framework for create rpm type spec files for solaris. Once you have one setup for asterisk then it is just a one line command to download and build asterisk. This is what I have been using to build asterisk on Solaris 10 for the past 3 years. It keeps the environment identical between versions. Have a look at getting that up and going first and then check out the spec file format and create one for your asterisk version you want to compile. My spec file is far from perfect at the moment, but it does work for what we require at the moment. Disclaimer: This is a little bit of work to setup and get working initially, but once it is setup and working, building subsequent asterisk versions and creating the Solaris SRV4 packages is a breeze J Thanks Bruce Hi Bruce, Thanks so much for that. I don't know what to tell you as to why I'm getting the error if you didn't. Maybe it's because I'm using OpenSolaris as opposed to Solaris? That's the only thing I can think of and Tilghman's comment also kind of hinted at that the Makefile and/or configure or the overall build process needs to be smarter to tell when the system is being built for Solaris or OpenSolaris. Also while searching for something else but a related issue, I found another thread that had talked about successfully compiling 1.8 beta on Solaris on Sparc. So there's definitely hope. But I think this might be an OpenSolaris thing as even though I don't have the sophistication of CBE and Sun Studio etc, I do have the reasonably convenient VM snapshots to get a clean system whenever I want to and I can tell you, there was NOTHING on this system other than a fresh OpenSolaris install, and the gcc-dev package. Hmm Anyway, let's see if the nice developers at Digium can find some time to put in a fix for this so the product might become buildable over Solaris AND OpenSolaris and people can then just go with the platform of their choice. Cheers, RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hello All, I have been banging my head against trying to get asterisk to compile on Solaris as well as OpenSolaris. I've tried to build various versions of Asterisk as on various versions of Solaris and OpenSolaris to no avail. Finally, I said, what the heck, I got the latest version of OpenSolaris that (pkg image-update) could get and then the latest ver of asterisk I found on the digium repo. Amazingly, configure and make menuselect went without a hitch, very clean. 'make' was going really well as well, in fact this is the farthest I've ever seen it ever go with the minor hitch compalining about format_mp3 but it suggested I use that script in contrib and download the code for that and that made it run again. BUT just my luck, it crapped out with this error *netsock.c: In function `ast_set_default_eid': netsock.c:250: error: structure has no member named `ifr_hwaddr' make[1]: *** [netsock.o] Error 1 make: *** [main] Error 2 * Can anyone please help me resolve this? I don't even know where to look. Google came back with nothing. Same with a search through the 30,000+ emails I have from the Asterisk mailing list only gave me the hint that it's a function from if.h which in OpenSolaris resides in /usr/include/net as opposed to maybe /usr/include/linux. Any ideas? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? Thanks \RR On Wed, Dec 8, 2010 at 4:43 AM, RR ranjt...@gmail.com wrote: Hello All, I have been banging my head against trying to get asterisk to compile on Solaris as well as OpenSolaris. I've tried to build various versions of Asterisk as on various versions of Solaris and OpenSolaris to no avail. Finally, I said, what the heck, I got the latest version of OpenSolaris that (pkg image-update) could get and then the latest ver of asterisk I found on the digium repo. Amazingly, configure and make menuselect went without a hitch, very clean. 'make' was going really well as well, in fact this is the farthest I've ever seen it ever go with the minor hitch compalining about format_mp3 but it suggested I use that script in contrib and download the code for that and that made it run again. BUT just my luck, it crapped out with this error *netsock.c: In function `ast_set_default_eid': netsock.c:250: error: structure has no member named `ifr_hwaddr' make[1]: *** [netsock.o] Error 1 make: *** [main] Error 2 * Can anyone please help me resolve this? I don't even know where to look. Google came back with nothing. Same with a search through the 30,000+ emails I have from the Asterisk mailing list only gave me the hint that it's a function from if.h which in OpenSolaris resides in /usr/include/net as opposed to maybe /usr/include/linux. Any ideas? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
On Wed, Dec 8, 2010 at 3:40 PM, Paul Belanger pabelan...@digium.com wrote: On 10-12-08 03:21 PM, RR wrote: Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: I'm in the process of bring up our remote Bamboo agents for Solaris, so I can see if I get the same issue. Which versions of Solaris and OpenSolaris are you using? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org Hi Paul, I haven't tried compiling it on Solaris 10 as yet, as OpenSolaris is a lot easier to update and download packages / dependencies etc. neverthess, the OpenSolaris version is: OpenSolaris 2010.05 snv_134b X86, running on a Core 2 Duo Quad machine inside a 64-bit Hyper-V VM. Let me know if you need more info. BTW, the way the OS was installed was through the ISO available on the OpenSolaris website and then updating it with 'pkg image-update' command and then following it with installing the gcc-dev package. Thanks \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.com wrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.org and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org G'day Tilghman, Thanks for that thread. I guess a few other things broke because of the change and the consuming application then needs to be a little smarter like you said (and suggested by darrenr) to detect whether you're on OSOL or Solaris. Does that mean I should check this same thing out on Solaris 10 as well and see what happens? I am so lost with the Solaris build environment as (and I whinged about this earlier too) there is no good way of obtaining the standard Solaris packages and dependancies and everything just goes all over the place and then one is left scurrying around to find where the damn library needs to be for it to compile. Anyway, I will open an issue and reference this thread and we'll go from there. BTW, THANK YOU for taking note of this and trying to help. You guys will have bottomless beer pitchers paid for if you guys help me get this working and are ever in the NY area :) Cheers, \R -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / Asterisk on Solaris
Hi Bruce, Thanks again for your generous response, please see a few comments inline On Sat, Dec 4, 2010 at 6:27 AM, Bruce McAlister bruce.mcalis...@blueface.ie wrote: Hi RR, Replies inline below *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RR *Sent:* 04 December 2010 01:17 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Zaptel / Asterisk on Solaris On Wed, Dec 1, 2010 at 3:58 PM, RR ranjt...@gmail.com wrote: Zaptel package isn't installing though ...crashes midway complaining that: *Operating environment requirement not met.** This package requires Solaris 7 or better. checkinstall script suspends* huh? I'm running 5.11, which according to some rigorous mathematical calculations, I concluded IS better than v5.7. Unfortunately, I've been away from the development world so long that I can't remember where to go about hacking a package and extract the scripts etc to change the logic or fix whatever is causing it to believe that my OS isn't meeting the min. req. Lastly, w.r.t to running it within a VM, yes, I do understand the timing problems etc, but this exercise is just to document how to compile Asterisk/Zaptel under Solaris 10/11 so when I do get a real Solaris machine, I have already sorted out all the issues with installing/compiling etc Thanks \R As of this writing, I have recreated my Solaris VM with the latest Solaris 10 U9 version and have managed to install and load the zaptel driver. This is from the SolarisVoIP but it must be a really old (haven't checked the version yet). Now, trying to go crazy here and compile the stock Asterisk 1.6.2.14 with it. -- I suspected it would have been changes/differences between OpenSolaris and Solaris. The packages at SolarisVoIP were built on the standard Solaris OS, as you found out J Try to compile version 1.6.2.15-rc1 as I had issues with a “timersub” routine on 1.6.2.14 that appears to have been fixed in 1.6.2.15-rc1 -- Ok, awesome. Will give 1.6.2.15-rc1 a try although in the beginning I did see compilation errors w.r.t to timersub routine, I don't see them anymore. I think it was complaining about some library that I then created a soft link for in a lib directory and that seems to have got fixed. But neverttheless I'll get the build you recommend. Question for the Digium dev team (if they bother reading emails from lowlifes like me): Are their special optimizations/options/conditions/checks ALREADY in place within the makefile/configure files that detect Solaris and if we want to go really crazy then detect 64-bit Solaris? Do I just fix my library paths with the LDFLAGS and just run configure or should I be doing something more, modifying makefile, makefile.rules, makefile.opts or the configure script itself?? -- The makefile already has some options specific to Solaris, however, I usually edit the makefile to include /usr/sfw which is where the standard ssl etc libraries are located on Solaris. The default makefile looks for them in /usr/local. If you also want to keep your application in the /opt tree then you will need to modify the installation path as well. I seem to recall an issue with ncurses or tr or something along those lines which made me include /usr/xpg4/bin in the beginning of my PATH so that it found the proper tool in one of the scripts. Other than that it should build cleanly on Solaris. With regards the 64-bit build, I’ve not tried it yet, but bear in mind that the 64-bit libraries for the likes of ogg/vorbis are not there by default in Solaris, most of the other standard asterisk library requirements are, you should, in theory only have to export libdir/64 to link in the 64 bit libraries. I’ve not tried to build a 664-bit version yet so I’m shooting in the dark here. -- Ok, right so this is where I'm having serious issues almost every step of the way. The problem is the stupid paackaging of Solaris and the difficulty in obtaining packages for them and their dependencies. Like for a week I struggled with figuring out how I could install/upgrade solaris 8 over the network to Solaris 10 with JUST a minimal core and the devlopment tools like gcc, gmake and some libraries etc. then I gave in and decided to just install the developers Metacluster since compiling/building Asterisk on it was more important to me right now. Then if I want to stick with purely the Solaris version of everything, my only option is to manually download packages I think I need from sunfreeware.com. If I use pkgutil or pkg-get, then I end up with the CSW packages and that will add to the complexity of PATHs to my libraries and binaries. Anyway, now that I'm done b**ching about Solaris (haha) you have hit on the core of my problem. It's the library paths that are messing me up. So this is how I ran configure: *LDFLAGS='-R/lib -R/usr/lib -R
Re: [asterisk-users] Zaptel / Asterisk on Solaris
Hi Tilghman, Thanks for your response. Please see my response below. On Sat, Dec 4, 2010 at 2:17 PM, Tilghman Lesher tles...@digium.com wrote: Changes that you need to make to get Asterisk to compile reliably on Solaris would be welcome reports on the issue tracker. What we have now are paths that worked for someone at sometime. Adding extra include and library paths should not cause problems, but what really should happen is that the configure script should be detecting the right paths and automatically adding them to the Makefile variables. Our standard build script for Solaris (1.8 and trunk only) does not alter the source at all, but does set PATH and LD_LIBRARY_PATH to ensure that certain values are included, and it does compile and run (and pass) our unit tests. So as mentioned in my response to Bruce, I did try to prepend these library paths to the configure script via the LDFLAGS variable. You're right about all the includes and library paths. The problem is, as I've mentioned earlier, for non-developer people like me, this isn't second nature and eveytime I have to do something like this, it's usually months or many times years since I'd last had to build anything from source and modify something anything in the code. So it would help if after these unit and sanity tests that are done for whichever build/release/version, if these can be added to the README or INSTALL notes for Solaris, for *that* build in a file like INSTALL.solaris or something. This won't happen everytime or for every release/, and I understand and appreciate that but it's a bit hard to get this going whent eh only resource we have are the pages from SolarisVoIP people which is almost 3-5 yrs old and I have not heard from Joe at thrallingpenguin after I sent him 2 emails about 3-4 weeks ago. We'll we more than happy to help in whatever way, if we can get a bit of hand holding and relatively fast turnaround/attention from the developers who can just point us in the right direction ... don't have to use up your machines, dev cycles or testing time, we'll do all of that if we can get help in troubleshooting the build environment. Anyway, coming back to the point, so are you saying that I should try trunk or 1.8 instead of mucking around with 1.6x versions? Sorry I have got back to Asterisk after almost 3 years, so haven't kept up with where I should be going and is it better to stick with 1.6x or just go to 1.8 as there's no upgrades or backward compatability requirements for me. Once I get this going, I promise to have an updated document uploaded somewhere or will mail to the list so someone can put it on the wiki. Thanks, RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / Asterisk on Solaris
On Thu, Dec 2, 2010 at 5:05 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: Assuming Solaris is anything like Linux, the installer will just be a shell script. Open the script in a text editor and search for the text of the error message. It will be wrapped inside an `if` statement, just alter this so the test always passes. I had to do something similar to allow the Flashplayer installer to install the 32-bit Flash binary into users' home directories held on a 64-bit NFS server and exported to 32-bit workstations, right from the server. -- AJS yes Solaris is a lot like Linux, well they're all just variations of the standard Posix-C old ATT Unix systems right? But the pkg files I have are just bundles like RPMs etc, and I haven't really explored how to open / extract the files inside a pkg and then muck around with them but I'm sure it's not difficult to do. I guess I haven't found out how to unpack a pkg file and extract the contents, then find the script that running and modify/edit it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / Asterisk on Solaris
On Wed, Dec 1, 2010 at 3:58 PM, RR ranjt...@gmail.com wrote: Zaptel package isn't installing though ...crashes midway complaining that: *Operating environment requirement not met. This package requires Solaris 7 or better. checkinstall script suspends* huh? I'm running 5.11, which according to some rigorous mathematical calculations, I concluded IS better than v5.7. Unfortunately, I've been away from the development world so long that I can't remember where to go about hacking a package and extract the scripts etc to change the logic or fix whatever is causing it to believe that my OS isn't meeting the min. req. Lastly, w.r.t to running it within a VM, yes, I do understand the timing problems etc, but this exercise is just to document how to compile Asterisk/Zaptel under Solaris 10/11 so when I do get a real Solaris machine, I have already sorted out all the issues with installing/compiling etc Thanks \R As of this writing, I have recreated my Solaris VM with the latest Solaris 10 U9 version and have managed to install and load the zaptel driver. This is from the SolarisVoIP but it must be a really old (haven't checked the version yet). Now, trying to go crazy here and compile the stock Asterisk 1.6.2.14 with it. Question for the Digium dev team (if they bother reading emails from lowlifes like me): Are their special optimizations/options/conditions/checks ALREADY in place within the makefile/configure files that detect Solaris and if we want to go really crazy then detect 64-bit Solaris? Do I just fix my library paths with the LDFLAGS and just run configure or should I be doing something more, modifying makefile, makefile.rules, makefile.opts or the configure script itself?? Wonder why no one responds to emails related to Solaris on Asterisk...even a search throughout the forum on the Digium website, hardly anything comes up regarding Solaris :( Can the developers PLEASE help? Thanks so much in advance, \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel / Asterisk on Solaris
Hi Bruce, Thanks for responding to my message. Doesn't seem like anyone is or is interested in running Asterisk on Solaris or if they are then they're being very secretive / quiet about it as I need a bit of help. Yes, I do know that SolarisVoIP people do have pre-built packages out there that I can simply install without having to deal with compiling them which I have actually done for Asterisk. Alas, the version of Asterisk they have in that pkg is v1.2.7.1. Also, don't know if you've recently looked at the page at SolarisVoIP but they do have a package for OpenSolaris v5.11 which is a fairly recent Solaris version however it beats me why it comes with Asterisk v1.2.7.1 instead of 1.4 or 1.6 even. Anyway, so yeah like I said, I have that installed, but I can't start it as it fails trying to look for Zaptel stuff. Zaptel package isn't installing though ...crashes midway complaining that: *Operating environment requirement not met. This package requires Solaris 7 or better. checkinstall script suspends* huh? I'm running 5.11, which according to some rigorous mathematical calculations, I concluded IS better than v5.7. Unfortunately, I've been away from the development world so long that I can't remember where to go about hacking a package and extract the scripts etc to change the logic or fix whatever is causing it to believe that my OS isn't meeting the min. req. Lastly, w.r.t to running it within a VM, yes, I do understand the timing problems etc, but this exercise is just to document how to compile Asterisk/Zaptel under Solaris 10/11 so when I do get a real Solaris machine, I have already sorted out all the issues with installing/compiling etc Thanks \R On Wed, Dec 1, 2010 at 1:10 PM, Bruce McAlister bruce.mcalis...@blueface.ie wrote: Hi RR, As far as I am aware the version of Zaptel on SolarisVoIP is out of date. Aditionally the versions of the packages compiled at SolarisVoIP are only available, as far as I am aware, for the Solaris platform and not the OpenSolaris platform, there may be subtle differences between the two that may be causing your build error. If you have a look at SolarisVoIP there are pre-built packages for SPARC/X86 hardware which you do not need to build yourself. In saying all of the above, your millage may vary with zaptel running in a VM as the timing is virtualized (via usb) and is not, as far as I know, very well supported within a VM. Thanks Bruce *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *RR *Sent:* 01 December 2010 00:55 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Zaptel / Asterisk on Solaris Hello nice people :) I have been struggling with trying to get Zaptel from http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained from the OpenSolaris Website. I have tried installing all the necessary packages, yet I keep getting errors no matter if I try using the gcc available at sunfreeware.com OR the blastwave CSWgcc packages and GNU 'gmake' (as suggested somewhere on the Internet). I have tried sending emails to the people at SolarisVoIP.com and To Simon, from Slimey.org who built/created this Zaptel Solaris Port, but it's been over 2 weeks and I've not heard anything from anyone. This is EXTREMELY critical for me to work...can anyone kind generous gentleman please help? Thank you so much \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel / Asterisk on Solaris
Hello nice people :) I have been struggling with trying to get Zaptel from http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained from the OpenSolaris Website. I have tried installing all the necessary packages, yet I keep getting errors no matter if I try using the gcc available at sunfreeware.com OR the blastwave CSWgcc packages and GNU 'gmake' (as suggested somewhere on the Internet). I have tried sending emails to the people at SolarisVoIP.com and To Simon, from Slimey.org who built/created this Zaptel Solaris Port, but it's been over 2 weeks and I've not heard anything from anyone. This is EXTREMELY critical for me to work...can anyone kind generous gentleman please help? Thank you so much \RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS in Asterisk on Solaris
No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TTS in Asterisk on Solaris
Hi Luis, Thanks for your comments. How / Why are you using that many TTS products? Do you have a preference of one over the other? Also, do you have any documentation / install/configuration notes that you might be willing to share re: your experience with Debian on Sparc and the TTS configuration you have. I agree with you. I will use TTS in its own native environment and have Asterisk talk to it using UniMRCP or something but I need a lot of help. Any help will be appreciated. Thanks \RR On Thu, Nov 11, 2010 at 11:22 PM, Luis Morales faston...@gmail.com wrote: I use Nuance, festival, Ibm tts and Loquendo. Now in your case, i suggest use tts on the recommend tts environment. Solaris is not standart system for tts products. Then you can plug tts system into asterisk platform. I use Debian for sparc and work excelent!! don't discard this option may be an good choice. Regards, On Thu, Nov 11, 2010 at 11:36 PM, RR ranjt...@gmail.com wrote: No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci
Re: [asterisk-users] TTS in Asterisk on Solaris
Sure, no worries. Will try that. What about advice on TTS setup. Would you have any notes on how best to setup high-volume TTS environment, like maybe a cluster of TTS servers and how Asterisk talks to those? Recommendations on how to set that up? I'm thinking about trying Festival/FLite and maybe Cepstral? How expensive is Loquendo? Thanks RR On Fri, Nov 12, 2010 at 1:35 AM, Luis Morales faston...@gmail.com wrote: Well, I use many tts products because i work with diferents telphone systems. Now for asterisk the best way for free is Festival and noon free is Loquendo. I'm not have notes to install debian on Sparc, i just only use debian readme :-) It's too easy, debian work for you :D Just download sparc image, burn it and install. Regards, On Fri, Nov 12, 2010 at 1:23 AM, RR ranjt...@gmail.com wrote: Hi Luis, Thanks for your comments. How / Why are you using that many TTS products? Do you have a preference of one over the other? Also, do you have any documentation / install/configuration notes that you might be willing to share re: your experience with Debian on Sparc and the TTS configuration you have. I agree with you. I will use TTS in its own native environment and have Asterisk talk to it using UniMRCP or something but I need a lot of help. Any help will be appreciated. Thanks \RR On Thu, Nov 11, 2010 at 11:22 PM, Luis Morales faston...@gmail.com wrote: I use Nuance, festival, Ibm tts and Loquendo. Now in your case, i suggest use tts on the recommend tts environment. Solaris is not standart system for tts products. Then you can plug tts system into asterisk platform. I use Debian for sparc and work excelent!! don't discard this option may be an good choice. Regards, On Thu, Nov 11, 2010 at 11:36 PM, RR ranjt...@gmail.com wrote: No, I want to use Solaris 10 on the Sparc platform. I've read a lot of reports and tests/benchmarks conducted that sow Solaris 10 actually performing better than all other Linux based Distros...not sure if that's been the experience of others in the group. I really want to know if someone has a high performance TTS based service running in a production environment. What product are they using as their core engine, does it handle and has available many different languages and can one build these independently of the telephony platform being used so I could use maybe Asterisk running on Solaris 10 and a cluster/farm of TTS servers for TTS processing. Thanks RR On Thu, Nov 11, 2010 at 10:21 PM, Luis Morales faston...@gmail.com wrote: You try install debian in your sparc platform ? On Thu, Nov 11, 2010 at 8:52 PM, RR ranjt...@gmail.com wrote: Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the core of the servers are Sparc based Solaris platforms. I found a group that seems to have done a fair bit of work on compiling Asterisk on Solaris, but I'm wondering if it'll be possible for me to have my core platform running Asterisk on Sparc Solaris and a set of Linux servers serving as a TTS cluster to which the calls can be thrown to for processing and then have them be played back to the user. Any ideas/advice? Thanks RR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello
Re: [asterisk-users] softphone with g729 codec
On 7/11/07, Guillermo Salas M. [EMAIL PROTECTED] wrote: On Wed, 2007-07-11 at 08:32 -0400, Maximo Villamayor wrote: you can prove this www.portsip.com You can use the older version of firefly that supports IAX2/SIP protocols and g729 codec. Get the sofhophone and codec from: http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/firefly-thirdparty.exe http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/g729.zip To enable the g729: 1.- Install firefly-thirdparty.exe; 2.- close firefly program; 3.- extract g729.dll from g729.sip to c:/program files/firefly; 4.- start firefly, setup a new account and enable the g729 check box; Regards, Gordon Henderson wrote: On Mon, 2 Jul 2007, jonny hashem wrote: Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? eyeBeam - the commercial version of X-Lite: http://www.counterpath.com/index.php?menu=Productssmenu=eyeBeam Gordon -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting Thanks guys! I have been silently watching this and doing my own bit of research on a freely downloadable softphone with g.729! I didn't find the quality of PortSIP that sh*t-hot but the Firefly version sounds good. BTW, does anyone know if these can both be used within a live service and freely distributed to our subscribers without legal/license implications on g.729 codec libraries? I wonder if the PortSIP SDK allows for the integration of advanced media engines like that from GIPS? I think that will bring about the quality into it but I wonder if that will lose the cost benefit that it gives in its native form. Also, anyone here know about multi-media web-based clients similar to what Gizmo plug-in does? I have readmany discussions by people from Mexuar and others but I want more something along the lines of the Gizmo Plugin. Any thoughts would be appreciated! Thanks all \R ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Session Border Controller time...
On 7/8/07, Dovid B [EMAIL PROTECTED] wrote: What does the NexTone run for ? - Original Message - From: Andy Brezinsky [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 03, 2007 8:17 PM Subject: Re: [asterisk-users] Session Border Controller time... We use NexTone for our SBC's on our network. We like: - 10,000 concurrent calls with media routing - SIP H.323 signaling with ability to take care of odd vendor specific issues - Basic routing engine allows you to create calling plans for individual end points - Limits by bandwidth or concurrent calls (or egress/ingress) for either discrete endpoints or via an iEdge group. - Easy GUI for those less tech savvy to do work on the machines. - Reasonable pricing on a per-port basis - Amazing Sales/Support teams. We've had some super funky requests we've thought about on a Friday night, they've got their teams together to walk us through every part of the configuration. Very knowledgeable and fun staff. (Seriously, best vendor support we've ever had, Hi Dan!) If you upgrade your SBC's to their RSM product you get basically a full Class 4 soft switch with a full LCR routing engine, reporting system and analytics engine. It's pretty powerful. Right now we're using just the SBC component and sending all ingress traffic to a egress trunk group (pointed to our OpenSER routers) but we're running a few thousand concurrent calls throught it. -- ~Andy Brezinsky On Tue, 2007-07-03 at 12:14 -0400, J. Oquendo wrote: Come on you carriers on the list... Give up the dibs what are you using and why? About to sledgehammer these SELECT * FROM GARBAGE WHERE SBC = 'nCite' Don't bother shooting me off Newport Networks stuff... Too pricey I agree with J.Oquendo! Maybe the story with 4.2 ver is different but their 3.1x line is horrible at the subscriber/access/line side, and they admit to it and have personally asked/recommended 'off-the-record' for me to go somewhere else for providing feature rich line-side features. A load of SIP METHODs/Messages aren't supported, no support for geographical redundancy (both SBCs must be placed physically in the same CoLo alongside with a x-over cable between them), Registration throttling doesn't work for me, neither does session-refresh, NAT traversal isn't adaptive (i.e. you can either media route everything or nothing, it doesn't detect that two endpoints might be behind the same NAT so don't bother media-routing them all the way to the PoP and back), doesn't load-balancing multiple application and/or call/proxy-servers (manually must assign priorities to each server) and many more but the worst and absolute worst is the support! I have solved more problems for them that I should be charging them for support instead of the other way around. I've found bugs, security holes, and incorrect implementation of the SIP RFCs. If the bug is obvious and they can figure out a solution for fast, they will work on it. If it involves investigation and/or major change/fix, they let it lie there. I had about 2 bugs that lay there in their system for almost 6 mths. Luckily I found workarounds for them and my service is running on those workarounds and will forever till we upgrade to 4.2 as we've been told that these might've (still no guarantee) been fixed in the newer releases. On the +ve side, their carrier side is good (but then, carrier/peering/interconnect is prob 25% as complex as the line-side) and robust, the quality is good and the pricing is very modularised, so you can cherry pick modules u want depending on what services you want to offer. Other SBC vendors sell you everything whether you ever use it or not! although I've heard now that Netrake has wised up and modularised their pricing after Audiocodes acquisition and having fired most of the original execs from Netrake. Anyone here heard of Covergence? I saw them at VON and had a LONG chat with them with a demo of their product. VERY neat, and am sifting through wads of their whitepapers before contacting them for inter-op for the Next PoP. Apparently the V-Dawg (Vonage) uses them not that that gives any credibility to anything but if anyone knows more than I do about them, please share! What about Acme Packet? Or Metaswitch SBCs, Juniper, Cisco, Sansay? Anyone written their own on Stacks provided by companies like Data Connection? oh BTW Dovid, You should be able to get very minimal config Nextones for about $30K/piece for about 2000 media routed calls and 20,000 registrations. This might've increased with the 4.2 train as you now HAVE to get the media-processor/DSP card which I believe is $6K extra HTH \R ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Localise VM_DATE timestamp like the voicemessage envelope
Hello, is there anyway or any plan to have the date/time stamp that's printed in an outgoing voicemail notification email to NOT be the date/time of the (*) machine but infact correspond to the timezone set for the subscriber under the TZ variable? I have the (*) machine set to UTC and when the notification email goes out, it prints out the date/time of the machine at which the voicemail was left but when you hear the envelope of the voicemail, it's the subscriber's local timezone. Which is ofcourse the correct behaviour. But this is not the same for the notification email. So, Is there a smart way of modifying the VM_DATE variable to read the DB to do what the envelope does? Perhaps a real smart DialPlan trick to pick that up during the time the voicemail is being left or something? If I were to use the externnotify, then how would I go about maybe ceating a script that can access the DB, get the subscriber's timezone, convert the machine's UTC time to the subscriber's timezone, and then create the same message? Just wondering if someone has actually solved this already and would like to help before I start to maybe writing a script of my own. many thanks, \R ___ --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] FW: Realtime Voicemail Password Change Not Working
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote: I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, enter new password ok, re-enter new password ok, password has been changed There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? I updated to asterisk 1.2.14 and add-ons 1.2.5 with no luck. I still don't see any update statement in the mysql.log when I change a password. I built a vmbox in the voicemail.conf file and can change that password just fine. Any suggestions? Thanks. JR Not sure about how you log in MySQL but using ODBC, in your odbcinst.ini or a similar file for Mysql, which keeps the settings for your db driver etc, you should be able to turn on logging. I can in odbcinst and it creates logs. The problem you have seems more like a permission problem however, the user you're using to log into the DB doesn't seem to have the permission to write to the table which keeps the user information OR the voicemail database itself. This problem becomes a bit trickier when your vm user table is actually a view of tables that hold subscriber/user information and is compounded by the fact if voicemails are being stored in a different db than where the sip/iax user information is being stored to derive sipusers and sippeers family values as then the user that asterisk is using to connect to the voicemail db will also need write permission in the db that stores user information. I dunno if any of that made sense but the password change works for me fine in 1.2.x as well as 1.4b3, haven't tested 1.4 Release yet. But in short, check pemissions for the user accessing the db(s) HTH \R ___ --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: Realtime Voicemail Password Change Not Working
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote: I use the same database for the sip, iax, exten and vm, different tables. When a sip device registers, asterisk writes to the database with updates to the sip table ipaddress, port and regseconds, so I don't think there is a write permissions issue from asterisk res_mysql to the mysql database. I thought of that also and changed the user to full access, but that didn't help. Mysql logs all database transactions in the /var/log/mysql.log file. I see all the query selects from the voicemail table and i see all the query updates to the sip table, but never see any query updates for the vmpasswd to the voicemail table. I would assume there would at least be errors if there was a permissions problem. I don't see where asterisk is trying to update the vmpassword through the realtime driver. How is your voicemail.conf file setup? Thanks. JR Interesting, well if you're seeing the other selects in the mysql.log then this update not showing up is bizarre. It would also mean that permissions are irrelevant if doesn't even attempt to change the password, as you'd rightly pointed out as well. I just tested it again and this is what I see in the odbc sql.log SQL = [UPDATE vmusers SET password=? WHERE uniqueid=?][length = 46 (SQL_NTS)] So it definately spits out something but my setup is considerably different to yours. I am using ODBC - FreeTDS - MS SQL Server for starters. There's nothing out of the oridinary in my voicemail.conf. What I do remember is some conversation sometime about the file locking fix that was put in or was being talked about regards to people using static files and multiple people trying to change their passwords. Just checking if you've compiled (*) clean without any mods to the code etc. I mean I have made mods to app_voicemail.c but nothing that affects passwords. Just for giggles, have you tried doing realtime update voicemail mailbox 1234 password 2345 ? I know you said that your db updates for regsecs, ip address etc is working but try specifically writing to your voicemail table and see if you are able to manually update the password. At least that way you can just focus on seeing why the password code is not being triggered in the (*) code when using MySQL. Sorry I cna't think of anything else to suggest at the moment. HTH \R ___ --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] prompt for send a message not played in VM main, HOWTO resolve
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: All, Just came across the prompt #3 from inside the top menu of VM in latest stable. Allison does not announce the prompt, but if you know it is there, you can press 3 successfully follow the prompts from there to send your message to other users on the system. But, of course, obviously, I am asking: how do I resolve the situation whereby the users are not hearing this prompt? (since most nearly all users will never know that this is here) (I sure hope my googling didnt miss this one) Thanks very much. Most appreciated. Jason Sjobeck Jason, I dunno if I understand your question properly. Did you not want the prompt to play or did you want the prompt to play? If it's the latter, then AFAIK, this has to do with the setting in your voicemail.conf file which allows users to send messages to other users, it's sendvoicemail=yes, if you turn this on, you'll hear the prompt. If it's set to no then you won't hear the prompt to allow users to send msgs to other users. If that';s what you were asking then your googling did miss it :P HTH \R ___ --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 registration
On 1/18/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, some body told me that you can make asterisk to register itself to another asterisk server. i just want to know whether it really can be done or not. i have googled a lot but no answeres. -- Regards Rizwan Hisham Software Engineer For servers A and B, You need to create a user a/c in say Svr A like rizwan with pwd 1234 and then in svr B sip.conf, put in a line register = rizwan:[EMAIL PROTECTED] You can now create a trunk that uses this a/c to SvrB to terminate calls there See http://www.asterisk.org/doxygen/1.4/Config_sip.html for more info HTH ___ --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] To 1.4 or not
Hello Gents, following on this discussion, anyone particularly have one view or the other about 1.4 and the voicemail and meetme enhancements (supposedly) it has? We're not in production yet, I've tested 1.2 up until 1.2.13 in the lab as well as 1.4b3, since none of them got a real hammering Its hard to tell at the moment if one is more stable than the other. Also, since I don't use it for anything BUT voicemail and meetme, would a lot of instabilities in the PBX side of things affect me? They shouldn't but who knows. Any comments and/or advice would be appreciated :) \R ___ --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] is it possible to use Asterisk voicemail as anouncement system only?
On 12/15/06, Michael Hamann [EMAIL PROTECTED] wrote: Hello, we are using asterisk in combination with the voicemail system. I´m just wondering if it is possible to switch the voicemail to an I am on holiday mode. This means that the unavailable message is played to the caller but no possability to record a message. So far I did not find an option in the voicemail.conf for this. Any ideas except creating my own ivr menu ? best regards Michael What happens if you use the maxmsg variable in voicemail.conf and set it to 0 or 1? Don't know if there's a minimum limit, the max is I think and default is 100. Maybe there's a built-in vacation mode feature but I don't know about it. HTH \R ___ --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] MeetMe Conferencing and Marked Mode
On 12/14/06, Tobias Wolf [EMAIL PROTECTED] wrote: Actually you don't need 2 different extension, but two different parameter-sets for the meetme-App. So, you have to implement some logic that detects, if the calling user has to be marked or not. It's your choice if you do this by dialplan logic or by AGI, or something else. The second PIN, which you can define in meetme.conf, is not for the marked mode, but for the admin mode. This gives the user some control over the conf. example: exten = s,1,playback(choose one for marked mode or two for normal mode) exten = 1,1,meetme(100,a) exten = 2,1,meetme(100,w) (Please note that the above is not an working dialplan ;-) ) Personally, I use an AGI for my Conferencing-Apps and let it generate the correct Parameters for the meetme App. Cheers, Tobias And what would someone have to do to sweet-talk you into sharing this AGI ;) ___ --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] MeetMe Conferencing and Marked Mode
On 12/14/06, Tobias Wolf [EMAIL PROTECTED] wrote:
Hmmm, there is really not much to share. Most of the code handles
Authentication or other stuff, like informing another server that a new
user has entered an conf-room, or updating databases.
Mostly I look an the CallerId to decide if this should be a marked
user (but there are not many scenarios ther the CallerID is known in
advance), or the Caller has to make the choice by himself, by touching
the right button.
I use Asterisk-Java as AGI-Implementation.
Everytime then an AGI-Script gets executed by an Extensions the
service-Method of my Java-Class will be executed:
private String adminMeetme = dMaXq; // not the 'a' for the marked Mode
private String userMeetme = dMXq;
public void service(AGIRequest req, AGIChannel channel) throws
AGIException {
// Ask if we want to be marked
char option = getOption(wantAdminPrompt, 12, 360 * 1000);
if (option == '1') {
// Yes
// This sets a Channel-Variable with the
// correct MeetMe-Parameters
exec(SET, MEETMEOPTS= + adminMeetme);
} else {
// No
// Parameter for a normal user
exec(SET, MEETMEOPTS= + userMeetme);
}
}
My dialplan looks like this:
exten = 1000,1,AGI(agi://localhost/askformarked.agi)
exten = 1000,n,MeetMe(${EXTEN},${MEETMEOPTS})
exten = 1000,n,Hangup()
This is a minimalistic Example, i have erased a lot of logic that has
little to do with the actual MeetMe-Room. But it is the essence of
dealing with the correct Parameters, there a lot of other way to
accomplish this. It depends on what you have in mind with your
application which way works for you.
So, you see there is no ready-to-use-multi-purpose-AGI which you can
simply plug-in to your Asterisk, sorry for that ;)
But i think the effort as not that great, even if you solve it only with
Dialplan-Logic, or AEL.
I am sure, you will come up with an solution :D
Take care,
Tobias Wolf
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Thanks Tobias,
That does help a lot actually, not least of all being saving me heaps
of time in trying to mess around with my dialplan thinking I didn't
know how to differentiate between marked and unmarked user. Maybe I'm
not thinking about all scenarios here but the logical thing (as per my
logic of course) would be to auto-set the user entering the admin
password as the marked user. The reason I say this is that let's say
you have a conference you've setup with a potential client, now you're
the host, but if he's the marked user he's the one without whom
the conference is pointless. So until he arrives, no one gets to talk
(I think that's how it works?). But if using what I'm saying, then the
host is also the marked user and as long as he/she's there, other
people can simply talk to each other and just wait for the potential
client to arrive. But if the host isn't there, then there's no one
to control/manage the conference hence all non-admin users should
simply stay in a holding pattern listening to MOH.
But I guess this discussion is only useful if the dev people are
reading this and they agree. Maybe I'm missing something, I don't
know.
Thanks for the AGI structure though, I had implemented this via
dial-plan except then it only works for a few static conferences with
static PINs. Our conferences reset conference PINs to random digits
every night and unless I do SQL queries (since I read meetme.conf from
the DB) I have no way of knowing what those PINs are and so can't
create DialPlan rules to check for the marked or unmarked user based
on the PIN. If I use your method, then I'll have to prompt the user if
they want to become marked or not. I don't want to offer the option.
But like you said, I'll figure out a solution (although I think I
already have while typing this) but something tells me, it'll be
difficult and messy without an AGI :)
Cheers
\R
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Re: [asterisk-users] MeetMe Conferencing and Marked Mode
On 12/13/06, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote: I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks I could be wrong but I reckon one way would be to give the host the admin password. You may or may not need to then add in your DialPlan the logic to mark the user entering the admin password as opposed to users who enter the general PIN. I'm assuming that since meetme is capable to authenticating against 2 PINs, it may auto-mark the user entering the password defined as the admin password in meetme.conf. HTH ___ --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 1.4b3 Realtime Voicemail
On 12/11/06, David Thomas [EMAIL PROTECTED] wrote: I only say two options for voivemail staorge when compiling 1.4, IMAP and ODBC. Are you using one of these? Are they configured? Does anyone know if Version 1.4 still does filesystem based storage of voicemail or if you must use IMAP or ODBC? David This seems to be fixed in the svn trunk r77. Seems to work now when I did a checkout from SVN. ___ --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: Asterisk 1.4b3 Realtime Voicemail
On 12/11/06, Martin Joseph [EMAIL PROTECTED] wrote: Sometimes if there is a message in a format that voicemail doesn't like, it crashes like that. Make sure the voicemail box is empty and try again... I have seen it crash like that with audio data it didn't like going back to before 1.2. Marty Yeah I thought it could be that but this was brand spanking new DB with no messages in it. It was only when I turned on verbose logging on (*) console that I actually saw what it crapped out on. It was do with non-definition of the odbc_request_obj function. So the bottom line is that you can't use voicemail realtime with ODBC with the stock 1.4b3 release unless you update all relevant files that are affected or just checkout the entire SVN trunk. A fair bit seems to have changed in the voicemail code alone since the 1.4b3 release ___ --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 stopped Matching Defined Peer
HI All, Something weird has happened to my (*) setup. Setup: I'm using a Realtime-Driven (*) server for voicemail which has the knowledge of all mailbox users on the softswitch which is remote to this (*) box. Since that's all this box is used for, all I have in the sip.conf is the definition of a peer (tried friend as well) which is qualified by its IP address. This is where the calls come to the (*) box from when the call needs to access voicemail. Peer definition in sip.conf Looks something like this [POP] type=peer host=xxx.xxx.xxx.xxx -- I have the actual IP of the originating peer here context=to-voicemail insecure=very disallow=all allow=ulaw dtmfmode=rfc2833 general part of sip.conf itself looks like [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 10.0.3.53; Address to bind to (all addresses on machine) externip = 60.xxx.xxx.xxx localnet = 10.0.3.0/255.255.255.0 nat=route disallow=all allow=ulaw then in extensions.conf I have, the definitions of extensions under 'to-voicemail context. This was working like a champ but all of a sudden has stopped working. I basically just get back a 407 Proxy Authentication message on my softswitch/proxy servers which I would think I shouldn't when I have a defined peer. It was quite happily printing out SIP debug messages which clearly stated Found peer POP, now I don't see that. I didn't change anything so I'm not sure why this is happening. And even if it is, what I can do fix it? Thanks \R ___ --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] Voicemail, SQL ODBC
RR, mate, I don't think that I have so many problems. 1.) I asked a simple question: Is it (still not) possible to connect Asterisk directly (= without ODBC) to mySQL for the purpose of storing voicemail data? Now, some posts later I've got a simple answer: No! Oh, haha sorry about that, I read these emails just to take a break from my regular work and filter them by keyword voicemail as that's all I use (*) for (and conference). Maybe I don't know enough about DB Connectivity by I thought the MySQL driver I was mentioning earlier is a direct connector to MySQL and doesn't need ODBC. ODBC I thought was for applications to talk to DBs for which there's no specific driver. So if instead of using unixodbc you compile with res_mysql (which you have for your CDRs) and then configure your res_mysql.conf with the DB info + in your extconfig.conf say something like voicemail = mysql,DSN,vm table it should work. But what do I know. Maybe someone can confirm this. 2.) It's not exactly clear to me why my extconfig.conf should humour you 1) it's just a phrase (i.e. humour me) and 2) Wanted to see if you're configuring your extconfig.conf properly, along the lines of what I said above 3.) You're telling me (and everybody else here) that you have *it* running with MSSQL. But you're neither telling what *it* exactly is or does nor *how* you made it running. Maybe you want your extconfig.conf post here? Umm *it* is the whatever the subject of the email and discussion is(?) and how I got it running is by what Derek just said :P. Ihave to use unixODBC, FreeTDS to get it to work with MSSQL server and store the voicemails in a DB. ___ --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] Custom Voicemail Notification Email
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: You can have your own external script to do whatever you want when vm is left from voicemail.conf: ; If you need to have an external program, i.e. /usr/bin/myapp ; called when a voicemail is left, delivered, or your voicemailbox ; is checked, uncomment this: ;externnotify=/usr/bin/myapp M Hi Marnus, externnotify, of course. I always end up spending months away from asterisk so by the time I come back to it, I've forgotten half the stuff. Thanks for the reminder. Now, maybe I'm stupid but how exactly do I get details to it regarding all those VM variables that are inserted when the email is normally sent out from voicemail. You know the VM_NAME, VM_DUR etc etc? I quickly tested this but as per the doco. for it, it automatically passes only 3 variables to the externotify script. Do I go parsing msg.txt file for the rest of the info? I may not have that in the case I'm using RealTime Voicemail. ___ --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] Custom Voicemail Notification Email
On 11/29/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
You could of course edit app_voicemail.c to pass more info...
Round about line 2329:
if (!ast_strlen_zero(externnotify)) {
if (messagecount(ext_context, newvoicemails,
oldvoicemails)) {
ast_log(LOG_ERROR, Problem in calculating number
of voicemail messages available for extension %s\n, extension);
} else {
snprintf(arguments, sizeof(arguments), %s %s %s
%d, externnotify, context, extension, newvoicemails);
ast_log(LOG_DEBUG, Executing %s\n, arguments);
ast_safe_system(arguments);
}
}
M
Right, I was looking at the sending email part to instead of sending
out the email, write a file with the relevant info and then I use
externnotify to pick up that file stick it into a template and send it
out.
Also, note changing the Content Type: text/html and recompiling has
allowed me to send and display emails as HTMLS. I'll just frigging
create an entire HTML page as one LONG string with pics and stuff and
give that a GO :)
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Re: [asterisk-users] Voicemail, SQL ODBC
And as I wrote before, Asterisk - mySQl connection is already up and runnig (for CDR). So it just would have been quick and easy if Asterisk could have used the same path for audio data. O.K., lets invest some time in installing ODBC. NOrbert Norbert, mate, I don't know why you're having so much problems. Do you wanna post your extconfig.conf here? just to humour us? I have it running with MSSQLServer a more complicated prospect than mySQL which has a dedicated driver for it, and it still works. ___ --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] Custom Voicemail Notification Email
Hello all, does anyone have a clever way of creating a customised email that goes out as result of the voicemail notification. And I don't mean Editing what you want in the emailbody, emailsubject, serveremail etc keywords. I mean custom in the sense that it has that info but the email is stylised to a certain format, with company logos and images etc. Is this more a sendmail question or do I need to do something within the app_voicemail.c code to make this to happen? Thanks, \R ___ --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] Voicemail, SQL ODBC
On 11/28/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Is the storage of actual voicemail messages in a database still limited to ODBC? If so, why? And is the use of mySQL and ODBC at the same time still a bad idea? If so, why? I want to store all of my voicemail stuff in a database so that I can give users web access to it, but I don't want to run web services on my * server itself. If it is all in a DB, I can have a web box and a separate SQL box and none of it should affect *. Mate, I can't say it with authority but I'm almost certain that the only DB that a specific driver was written for is MySQL. I think if you use res_mysql.o you should be able to talk to mySql directly without needing ODBC. I'm using ODBC via FreeTDS to an MSSQL Server, now that might be a disaster waiting to happen but won't know till I put some load on it. It seems to work well in the Lab with about 4-5 asterisk servers ALL accessing voicemail concurrently. ___ --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 and FreeTDS 0.64 or 0.63
Hello all, just curious if anyone's successfully compiled (*) with the latest FreeTDS code/driver. The Makefile in (*) seems to only take care of 0.63 or older. I tried to muck around with it a bit into tricking to compile for not just 0.63 but anything later than 0.62 but it seems to crap out complaining about CDR modules, which I really don't need. It's been a while since I tried it but I seriously doubt there's any dev. done to focus on intergarting MSSQL or non-open-source DBs with (*). If someone's done it or knows how to do it or even can tell me if it's even worth it, then I'd really appreciate your comments. Best Regards, \R ___ --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] Reading Voicemail Config from MySQL [+ ODBC]
On 11/6/06, Mosiuoa Tsietsi [EMAIL PROTECTED] wrote: Hi, After some more searching I decided to try USING unix ODBC for the connection. I have both the unixODBC and unixODBC-devel packages on my fedora box: [EMAIL PROTECTED] /]# rpm -qa | grep -i unixodbc unixODBC-2.2.11-7.1 unixODBC-devel-2.2.11-7.1 Here are my odbcinsi.ini and odbc.ini files respectively: [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc.so Setup = /usr/lib/libodbcmyS.so FileUsage = 1 --- [MYSQL-asterisk] Driver = MySQL Description = Data source for dynamic asterisk voicemail configuration Trace = Yes TraceFile = stderr SERVER = localhost USER = root PASSWORD = rootroot9 PORT = 3306 DATABASE = asterisk - Below are my res_odbc.conf and extconfig.conf files for supplying details of the DSN name and and database/table for asterisk [mysql1] enabled = yes dsn = MySQL-asterisk username = root password = *** pre-connect = yes --- [settings] voicemail = odbc,mysql1,users --- I am able to execute: [EMAIL PROTECTED] /]# isql -v MySQL-asterisk +---+ | Connected!| | | | sql-statement | | help [tablename] | | quit | | | +---+ SQL which shows I can connect to the database on the command line using my DSN name. In the asterisk CLI however, the command: asterisk*CLI odbc show No such command 'odbc' (type 'help' for help) fails which is supposed to show connections to MySQL from the CLI. ANd lastly the command: asterisk*CLI realtime load voicemail mailbox 7521 No rows found matching search criteria. Nov 6 00:33:10 WARNING[2965]: config.c:920 find_engine: Realtime mapping for 'voicemail' found to engine 'odbc', but the engine is not available also fails. Where are I going wrong? Thanks. Mate, doesn't sound like you have the res_odbc.so module loaded. Make sure in /etc/asterisk/modules.conf you have a load = res_odbc.so or on the CLI type load res_odbc.so and then give it a whirl. ___ --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] My Phone Review- Large Scale Corp Deployment.
On 11/8/06, Steve Edwards [EMAIL PROTECTED] wrote: All calls come in from a Tekelec 7000 via SIP. Out of a peak of 200 calls, probably around 100 are in meetme, others are listening to recorded messages or bouncing around in the menus. Sounds exactly like what people in my system would be doing. No OS tweaks, no Asterisk source tweaks. A TE410p is used as a timing source. The sound quality was not acceptable with ztdummy. Aha, so that's something I don't have and most prob. can't have (no empty PCI slots left on the 1U servers). Hmmm maybe that might make the difference between how many conferences my boxes will handle before it starts to sound bad! I stripped down /etc/asterisk/modules.conf just 'cause parts left out don't get broken :) Agreed, I have even removed non-used conf files so the size of (*) in memory is significantly smaller. My sip.conf only allows ulaw, but show channel shows some using ulaw and some using slin. This may be changing as the calls bounce from meetme to recorded wav messages. The Zap pseudo channels show ulaw -- I would have expected slin. Somebody who understands codec switching could help out and explain it to both of us :) Think you would only see slin if some system playback needs to access non-ulaw encoded files or users come on a different codec than others. Since the latter isn't happening, there's no need for your system to convert anything to slin, which is why your systems shows the Zap pseudo channels as ulaw and playback of recorded messages doesn't use the Zap pseudo channels. So unless my understanding is wrong, what your systems shows is consistent with your description of the settings you have there :) top refreshing every 3 seconds shows the asterisk process consuming from 10% to 70% of the CPU. top refreshing every 30 seconds shows around 30%. Does anybody know what causes the spikes. Yeah I'd be interested to know as well. I wonder if creation/tear-down of sessions does that. A conference in session should eventually get to a stable CPU consumption. You might want to have a test system and either through sipsak or manually create a bunch of conferences and watch the CPU. If you're playing the entry/exit sounds, recoding and announcing names, playing participant counts and all of these are non-ulaw encoded prompts etc. you will get those spikes as that's where codec-translation will happen. ___ --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] My Phone Review- Large Scale Corp Deployment.
On 11/8/06, Steve Edwards [EMAIL PROTECTED] wrote: I have alaw, g729, gsm, ulaw, and wav sound file sets so that should cover the transcoding bases pretty well. It should but if you're not allowing anything but ulaw all of those are probably not ever being used. What you might want to have is slin encoded prompts, which you didn't mention if you have or not. Although not sure if even that will help in cases where, say If you allow people to record their names which are then played upon entry and exit of a participant of the conference, I would think they're recorded in slin format (it being the default format for (*)), there is your codec-xlate right there. I wonder if there's some place where we can control what codec format is used to record these names and stuff in meetme. If this can be controlled and we can standardize the entire thing with ulaw, that should help a LOT, I'd think. And creation/termination of conferences could also be causing CPU usage, all adding upto the spikes. Butwhat do I know, maybe a digium person can shed more light on 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] My Phone Review- Large Scale Corp Deployment.
On 11/2/06, Eddie Johnson Jr [EMAIL PROTECTED] wrote: Hello Matthew, Did you test Snom or Sipura hard ip phones? I was considering Budgetone for an office of 10 users. After reading your testimonial I will have to re-think my selection. FWIW, after having played with 3-4 BudgeTone phones on 3-4 separate occasions, out of which 2 actually just died the very next day (came to work to find them, again on seperate occasions, with LCDs cleared out with greenish-blue tint, the speaker light lit and No Tone), that's when I concluded that the BudgeTone is surely Budget but No Tone! ___ --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] My Phone Review- Large Scale Corp Deployment.
On 11/2/06, Steve Edwards [EMAIL PROTECTED] wrote: I'm running CentOS 4.4, Asterisk 1.2.13 on HP DL380's. My application is mostly meetme conferences being created and closed all day long. Peak load is around 200 SIP calls. I was crashing 7 to 10 times a day until I booted a non-SMP kernel. I haven't had a crash since. Meetme does not play well with SMP. HI Steve, I have constantly got conflicting reports about meetme and can't really make up my mind to actually put meetme into service till I find something better or just stick with meetme and be happy? I like the features it has but performance wise I have heard all sorts of things, yours being the most positive so far. so just wondering if I can learn something from you. So, is there anything special you've done in terms of configs, modules, OS tweaking/tuning and the like, in other words, anything over and above simply installing OS and (*) with meetme for the system mentioned above? Have you standardised codecs across the board to minimise translation overhead? If so, then what codec are you using? Are all your users on IP or some can come through the PSTN via DIDs etc? Thanks Ranj ___ --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] G726 prompts
On 10/3/06, Jay R. Ashworth [EMAIL PROTECTED] wrote: It seems unreasonably difficult to get a list of the supported formats, but does sox (http://sox.sourceforge.net/) do what you need? Cheers, -- jra hey Jay, thanks but I am not sure what to tell sox as my output format to be. I must admit, I missed it the first time I was thinking about using it. Should've looked at the man page. This time I looked at it again It seems like I could convert ulaw pcm files into the adpcm format. But what output format do I choose? I tried raw but that doesn't work. I tried using the following like sox -r 8000 -c 1 input.ul output.raw anyone know what the correct parameters are for using sox to convert pcm or ulaw prompts into g726? thanks \R ___ --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] G726 prompts
Hello All, does anyone happen to know of a good utility or CLI tool to convert prompts into a g.726 format? I tried using the convert utility in (*) but it doens't like G.726. I understand I can just hunt around the net for it, but if someone knows one off-hand that I can run on linux and even run it inside a script that would be great. Cheers, Ranj ___ --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] Why not g726-32?
That's all well and good, but there are some phones out there that pack samples into RTP payloads using the AAL2 direction. This causes interop nightmares (i.e. your phones talk G.726-32, someone elses phones talk G.726-32, but it sounds rubbish when you attempt a conversation). I would guess that this might be why people avoid the G.726 codec. Interesting, maybe the reasons you and Rich stated might be some of the reasons I suppose. Thankfully neither of these will affect us since all the voip gateways/IADs and phones will be distributed and certified by us and BYOD type of a scenario will be highly discouraged PLUS I'm thinking of using g726 only when people want to interact with *. Every other time they'll be using g711 or g729 for off-net calls. This topic is still open, if anyone else has some interesting comments about 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] Why not g726-32?
That's all well and good, but there are some phones out there that pack samples into RTP payloads using the AAL2 direction. This causes interop nightmares (i.e. your phones talk G.726-32, someone elses phones talk G.726-32, but it sounds rubbish when you attempt a conversation). I would guess that this might be why people avoid the G.726 codec. Interesting, maybe the reasons you and Rich stated might be some of the reasons I suppose. Thankfully neither of these will affect us since all the voip gateways/IADs and phones will be distributed and certified by us and BYOD type of a scenario will be highly discouraged PLUS I'm thinking of using g726 only when people want to interact with *. Every other time they'll be using g711 or g729 for off-net calls. This topic is still open, if anyone else has some interesting comments about 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] Why not g726-32?
On 9/16/06, Rich Adamson [EMAIL PROTECTED] wrote: RR wrote: All, is there anyone who uses g726-32 ? If not, then does anyone know why don't people use it? I use g726 on iax links between systems and to teliax.com for LD calls. Have no idea if its -32 or what though. What ships with asterisk (in terms of g726) has been working very well for us with the exception of a period of time where all g726 calls via teliax were not usable. Teliax had to have had a problem or was playing around as that was the only iax link that had bad audio. Thanks Rich for the positive email about g726. Just FYI, (*) supports only g726-32 AFAIK so that's probably what you've been using. I don't have the worry of Teliax as I'd probably never be using them or at least not in the immediate/near future. Like I said, all I want to do is avoid usage of license fees, save bandwidth, and not stress out my systems with cpu intensive codecs like g729 and maybe find something that can still deliver comparable quality. I'm still confused as to why more people and carriers don't use g726 however. Anyonbe else can shed any light on 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] voicemail access thru apache on another server
have a look at Wiki for asterisk + odbc storage. The database for storing entire voicemail messages can be stored on a local or a remote database. Then you can do whatever you want with it. You will have to recompile asterisk by turning on ODBC storage. It's all there on the Wiki ___ --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] Why not g726-32?
All, is there anyone who uses g726-32 ? If not, then does anyone know why don't people use it? It's free, and provides the best compromise on quality, bandwidth and cpu load (judging by it's specs and algorithm) and oh did I mention, it's FREE? So why don't people use it? Any ideas? Is it too good to be true or it's not what it sounds like? Oh and since I am only looking at codecs to use between the subscriber and our system (no carriers involved), the popularity and ubiquity of g729 and g711 aren't a qualifying factor for this particular discussion :) Thx \R ___ --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 HIGH AVAILABILITY
You can achieve the active-passive setup using Linux HA techniques using heartbeat and the like. You can also do load-balancing with LVS. What I am not sure about is maintainance of call and session states between the two servers such that when one server dies, the other server picks its IP addresss, it keeps the calls in progress, up. There might be ways of sharing call states using Asterisk Manager interface where each server intermittently logs into the other to share their data OR if this info is temporarily stored in the astdb then maybe you can share the same astdb between the two servers. I am not sure about (*) internals enough :( But this should give you some ideas, I hope. ___ --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] Capacity for transcode G711 to G729
Hi matt, sorry this might be a stupid question but is a bit pertinent to me, I'd asked something similar in one of my last email regarding SMP. Do you know if (*) is capable of making use of HT support i.e is multi-threaded and improves performance for operations like transcoding? Is that a valid question or is this only dependant on the OS/Kernel, the CPU itself and the chipset on the motherboard? If I boot into an SMP kernel with Asterisk compiled with the SMP kernel source, would it just make use of multi-threading as the load increases on cpu-intensive operations? Also, when you said the normal is 120 simultaneous transcoding operations, what is normal? I have a P4 w/HT 3.4Ghz, 2GB RAM machine. Would that be above or below normal? Thanks much \R I'd guess at around 200-300 absolute max if the calls are spread evenly across CPUs. Normal is around 120. ___ --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] Prompts playback changing tempo w/ SMP kernel
Thanks Leo, great explanation. Will do some additional research and
try out a few tests if I can find the time to setup a small load-test
sort of a scenario but it does sound from your explanation that
symmetric multi-processing is what we need to share the load and get
double or close to double performance. Scheduling by its very nature
wouldn't be multi-tasking but rather a way to use up idle times to
perform more tasks and I don't believe there would be an idle
time/wait-time between a cpu-intensive task like transcoding in which
it'll get the time to use the other logical cpu to run another
transcoding operation. And the only thing I can think of why people
might be suggesting to turn HT off is because it has overhead and why
put up with it if we're not using it? Is that the thought behind it?
Anyway, thanks again. Looks like No VM and No HT for me :-{
\R
On 9/8/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Most virtualization platforms don't guarantee accurate timing. It's a
fundamental implementation issue. For example, if your host OS is
capable of 1000Hz. You won't expect the client OS to be able to do
1000Hz due to the overheads. Best case might be 999.9Hz, worse case
could be anything below that. IIRC, Xen VM used to do only 100Hz.
Asterisk is multi-threaded so SMP (either multi-way or multi-core) would
definitely help. But, HT would not. HT is still essentially still 1 CPU.
The CPU appears as 2 CPU by taking advantage of wait times to
multi-task. Good for normal usage like word processing, but bad when
running CPU-intensive tasks like codec transcoding.
___
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Re: [asterisk-users] Capacity for transcode G711 to G729
Hi Matt, The best use I have seen is the newly converted IAX2 which can use multithreading in version 1.4, the beta of which should be released later this week. The best idea would be to compile Asterisk, run some tests (show translation recalc 60) with HT turned on, restart the box, bring it up with HT turned off and try again. What's the best way to know for sure that you've everything setup the right way to use HT with Asterisk? There're so many things, I'm not quite sure if I am turning or conversely not turning enough things on/off. I do the following right now: - in /usr/src, I have the symlink of linux and linux-2.6 pointing to the location of the src of the smp kernel like e.g. /usr/src/kernels/2.6.9-34.0.1.EL-smp-i686 - Then do a fresh 'make' on asterisk with these symlinks in placehe - Then reboot and turn on HT in the BIOS - Then reboot with the smp kernel Is that it? If I compile with the linux/linux-2.6 symlink pointed to the kernel of the NON-smp kernel, then reboot in the non-smp kernel but leave HT turned on in the BIOS, does it matter? would that be enough? or should i turn off HT in the BIOS as well to avoid it causing issues? In my experiments with using (*) inside of a VM and doing SMP I'd seen that simply booting into an smp kernel gave me timing issues even when (*) was compiled against a non-smp kernel source. I don't see these problems on a real machine but that's just one call. Who knows what'll happen if I throw 100 at it. Would love to see the results of this test you're setting up. At the expense of bandwidth, maybe I'll just stick with g711 all the way through and save money on g729 licenses and load on my machine. Any thoughts on g726? Would using g726-32 be a good compromise on bandwidth and cpu power instead of g711 or g729? Thx \R ___ --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] Experiences, Tips on Voicemail storage using ODBC or IMAP?
I am currently running this with UnixODBC - FreeTDS - MSSQL Server 2K ( please don't hate me for using an 'evil empire' product amongst the pure sanctity of open source :D). But the results are, well...So far so good. But I can't say much because the most i've tried is 4 concurrent connections to the DB for users trying to access their voicemails but it does well. All I do is voicemail and conference so my extensions.conf is literally 20-30 lines. Would love to put a few hundred users come in to see what breaks first. Would also love to hear other people's experiences. ___ --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] Prompts playback changing tempo w/ SMP kernel
HI Mojo, thanks for that. Sounds like a hidden option. It doesn't show up when I do a tab after I type show translation on the CLI. But to respond to your comment, I thought that's what it was, as in calculated based on the current load of the system but the fact is that there is absolutely NOTHING running on the system. It's an absolutely clean install of minimal linux and (*). Watching top shows either 0.00 in all three columns or sometimes 0.01 to a max of 0.03. The change in the translation times is varying far too much if I stop and start asterisk consecutively say 5 times with just typing show translation after the start, then stop it, then restart and type show translation again. This just sounds too weird. Maybe the (*) developer can figure this out, but I doubt they're reading 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] Prompts playback changing tempo w/ SMP kernel
Hi Leo, Sorry mate, I thought I had done some research but I only found one reference to it somewhere and it stated that if you have more than one VM running inside VMWare server v1.0, then there are timing issues where the clock seems to vary randomly. I figured that didn't apply to me since I had the latest version and I was only running 1 VM. I should've used a better search criteria maybe :( The only way I discovered it was the VM was by actually getting one of my HT machines and installing (*) on it with SMP and it works fine natively. Just VMWare server seems to have timing issues. Again, sorry for that. BTW, do u happen to know if it's even worth my while to use SMP? I've read conflicting reports. I guess it would depend on (*) being multi-threaded or rather chan_sip being multi-threaded? is that the case? Thx again, \R On 9/7/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Ranj, Sigh :(. It would have save you and us a lot of time if you'd mentioned this fact earlier: Oh also, note that this system is running inside of a Virtual Machine with 768 RAM and a 3.4GHz CPU although NO other VM is active on this VM server. When running in a VM (like VMWare), the timing is not guaranteed. Search the archive. 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] includes in realtime ??
Ben, The family name is not sipuser, its sipusers. So try this command realtime load sipusers name username and see if you get nothing. What about? realtime load sipusers username username ? To answer your question, any change in the tables holding this sip users information comes into affect immediately. That's the whole point of realtime :) Cheers, \R ___ --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] includes in realtime ??
Ben, that's exactly how it is, the load command is only for you to see what's being pulled from the database and to test if realtime has been configured properly. If you see nothing, then I suspect realtime for you isn't really working and the calls that are working are being looked up in the local conf file. You might have to start doing some toubleshooting. What does your extconfig.conf look like? You might wanna post it here. Also, remove or comment out any extensions related info from sip*.conf files. What's the output if you type: asterisk -rx sip show settings | grep -i realtime on the linux command line? Lastly, ensure there's no errors logged with regards to connectivity to the database. Many pieces need to be in sync for it to work properly. I use it with UnixODBC - FreeTDS - MS SQL Server and it works beautifully :) If you're using a local MySQL database, it should be a piece of cake. Check you're loading the res_mysql module, check for config issues in res_mysql.conf and ensure yur user has permissions to access your asterisk database. Hard to suggest how to do all that without knowing ur exact setup. Sorry, the best I can do for now :) Goodluck \R ___ --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] includes in realtime ??
Assuming you have the tables as named int he extconfig.conf as well as the database astDB, how about enabling the module app_realtime.so? Also, if you're using mysql, I don't think you need res_odbc, res_config_odbc. Instead try turning on app_realtime.so and pbx_realtime.so and see how you go :) ___ --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] includes in realtime ??
I use rtcachefriends=yes and any changes I make in my database become effective immediately along with also getting the MWI functionality. Even though what you say makes sense. Go figure! Ben, yeah if it shows it's loaded then it's there for sure. Sorry I asked for it as in your module listing there wasn't any of these modules. I'm at the end of the rope on troubleshooting your issue. Maybe more detail is needed. Esp when you're saying that your sip.conf general section has just two entries. Where's the rest of it, not that a lot needs to necessarily be there if you're not doing anything too tricky. But I would go with removing the rtcache command from the sip.conf file and try and get realtime working in realtime, if that doesn't sound too whacked, just in case it's working off of some cached data, which is why your old codec selection seems to still work even after you change it. Have you looked in your asterisk log file (full) to see if its complaining about errors when you do a realtime load command? The only time my realtime load comes back empty is when it's got a permission problem of some sort on the DB side and one time it happened because of some delay that was introduced coz of some heavy logging or something, don't quite remember it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Prompts playback changing tempo w/ SMP kernel
Hi all, (2nd attempt) this is probably a weird question and something I'm not doing right but I got this bizarre thing going on here. When I boot the system with the SMP kernel and compile (*) with the smp kernel source (actually even if I don't compile, but as long as I boot into the SMP kernel), I get this problem where calling into the system, say to check my voicemail, the prompt playback continously changes tempo. The prompts are played in slow-motion, and then it speeds up to its normal speed, then goes back in slow-mo and so on. It happens (I think) at constant periods. Only the tempo changes, not the pitch of the prompt. Does anyone have any idea what could be happening? I have watched topconstantly but haven't noticed anything bizarre in terms of CPU or Mem usage. This is on a 100mbps LAN with nothing much else on it. And it only happens when it's booted into the smp kernel. So it's something to do with smp, thread scheduling, or some buffer BUT I don't know what exactly. All you champs out there, esp. the asterisk-dev people, any light you can shed on this? Thanks much \R ___ --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] Prompts playback changing tempo w/ SMP kernel
Hi Zoa, thanks for responding. Ok, now where do I find this? I'm running 2.6.9-34.0.1 kernel. I tried doing a bit of search and it seems like that the ability to change the frequency doesn't appear till 2.6.13. Am I looking at the right thing? Any hints? ___ --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] Prompts playback changing tempo w/ SMP kernel
Hi there, sorry I wasn't sure exactly where to start so didn't know what info to provide. Now that I know, here's the info 1) using a P4 w/HT 2) Using CentOS 4.3 with the 2.6.9-34.0.1-smp (Note, this was installed through an rpm, but the (*) and zaptel code is being compiled against the source of this) 3) I have tried it with and without ztdummy, and nothing changes. Although voicemail should have nothing to do with ztdummy, am I correct? 4) I have also tried with and without uncommenting the line for GSM optimisation for MMX processors line in the Makefile 5) I've also tried rebooting the machine with the line acpi=ht at the Kernel command line 6) Also tried strictly using one codec so as to avoid transcoding to see if that was it 7) I've tried booting into an SMP kernel without building (*) and zaptel for an smp kernel None of the above has helped. If I don't boot into an SMP kernel at all, it works fine. Also, at every start of (*), the show translation command shows different transcoding times without changing a single thing in the system in the way of config etc. Why is that? Oh also, note that this system is running inside of a Virtual Machine with 768 RAM and a 3.4GHz CPU although NO other VM is active on this VM server. Any ideas? Thx \R ___ --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] Prompts playback changing tempo in SMP kernel
Hi all, this is probably a weird question and something I'm not doing right but I got this bizarre thing going on here. When I boot the system with the SMP kernel and compile (*) with the smp kernel source (actually even if I don't compile, but as long as I boot into the SMP kernel), I get this problem where calling into the system, say to check my voicemail, the prompt playback continously changes tempo. The prompts are played in slow-motion, and then it speeds up to its normal speed, then goes back in slow-mo and so on. It happens (I think) at constant periods. Only the tempo changes, not the pitch of the prompt. Does anyone have any idea what could be happening? I have watched topconstantly but haven't noticed anything bizarre in terms of CPU or Mem usage. This is on a 100mbps LAN with nothing much else on it. And it only happens when it's booted into the smp kernel. So it's something to do with smp, thread scheduling, or some buffer BUT I don't know what exactly. All you champs out there, esp. the asterisk-dev people, any light you can shed on this? Thanks much \R ___ --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] REGISTER attempt
Also, keep in mind that from what I had understood, Vonage required any endpoint/acct. to register with them every 30 secs (I'm assuming they set the register expiry timer to 60) to ensure all endpoints keep their firewall pinholes open. This just got proven for a fact now that they do this i.e. require a reg. refresh every 30secs. ___ --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] Can the codec/format for name/greeting in voicemail be changed?
Sorry to badger everyone on the list but I never heard from even a single person on this so felt maybe I'll repeat it, just in case, it got unnoticed. Any ideas if it's possible to either record greetings/names in a different format than GSM OR be able to convert these voicemail subscriber greetings in my database to some other format? This is if I'm storing the voicemail and all greetings/etc in a SQL Server using Realtime. Thanks so much \R ___ --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] Can the codec/format for name/greeting in voicemail be changed?
Hello people,
before I go hunting on Wiki and Google, if maybe someone here knows
the answer to this.
This is in regards to the voicemail system. Is it possible to change
the default/native format in which the greetings and outgoing messgaes
for a user's mailbox are stored? It seems like (*) records everything
in a GSM 6.10, mono 8kHz. If I was using the filesystem, then I could
run a cronjob or something and convert all greetings etc. in the
formats that I expect the endpoints to be using, but since I use
Realtime, I don't have that luxury.
How then can I get (*) to either record in a different format OR be
able to convert these voicemail subscriber greetings in my database to
some other format?
Any ideas? suggestions?
Thanks in advance,
${RR}
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Re: [asterisk-users] Cisco Router QOS and IAX2
Bruce, this might be able to help give you some hints or a place to start: http://www.voip-info.org/wiki/view/QoS+Cisco Hope that helps \R ___ --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 set externip in sip.conf automatically?
Larry, am I missing something but you seem to be putting the externip
into the MYIP variable but reading some EXTERNIP variable through
$ENV{}. Shouldn't you be doing something like externip=${ENV{MYIP}}?
The other issue is also the use of curly brackets as opposed to
paranthesis. The snip from the manual seems to use curly brackets but
you're using paranthesis in your example above.
Just silly things to watch out for :D
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Re: [asterisk-users] Prompts recording for Asterisk
Nitin, I'm sure others have better advice but there's no best format per se. Whatever makes asterisk and more importantly the CPU work less in playing those prompts is probably best. from what I understand (*) picks up the best suited format based on the capabilities of the channel and endpoint. If you have endpoints that connect using different codecs, you'd want to have the prompts in all of those formats on your machine and (*) will pick up the relevant ones thus avoiding transcoding. You can find all information on this page: http://www.voip-info.org/wiki-Asterisk+sound+files. What I'm going to do is have the prompts be recorded in a wav (44khz) and then downsample them to 8kHz 16 bits windows wav. Then use the Asterisk 'convert' utility to convert all prompts to all diff formats I expect people to use. Hope this helps ___ --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 set externip in sip.conf automatically?
Mate, I'm beginning to think that it can't be done. As in, maybe
you're not allowed to put anything into externip other a valid IP
address and the $ENV{} variable doesn't really work there. You might
want to decipher your externip by registering your server with a
dynamic dns service and then lookup your IP through an nslookup
periodically. Then do some sort of a check and if the address has
dynamically changed, then rewrite your sip.conf file and do a CLI 'sip
reload' or 'restart when convenient'. Not sure why your IP address
should change that frequently anyway, so the approach I mentioned
should cover you. Maybe there are better suggestions out there.
BTW, In the newer versions, maybe it's in 1.4 only, you can use the
keyword 'externhost' where you specify the FQDN of the server, and it
will then lookup your external ip all on its own. You can then use
externrefresh to tell (*) to look it up every so often. Not sure what
the status of this feature is in current 1.2.x releases
Good luck
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[asterisk-users] 1.2.10 and 1.2.9.1
Hello good people, I'm sure this has been brought up previously but I basically wanted to wait to resurrect this topic till 1.2.10 has been out for a little while, like a cpl of mths. Now I think it has and I just wanted to request for peope who've chosen to upgrade their systems to 1.2.10 to provide their opinions (whomsoever chooses to provide one) about its stability and/or bug fixes as opposed to 1.2.9.1. I'd read a lot of mails about people having upgraded to 1.2.9.1. only to realise that they were better off with 1.2.7 or 1.2.6. Has this been the case with 1.2.10 or is this definately a more stable release specifically with regards to voicemail w/realtime and MeetMe. Thanks in advance to all who respond :) ___ --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_conference
Yes ___ --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_conference
Follow the instructions here: http://www.voip-info.org/wiki/view/Asterisk+app_conference There's no config file where conferences are stored. You need to add them to astdb using the 'database' CLI command like so: database put conferences 1234 9 Look at the setting up conferences section in the Wiki ___ --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
