[Asterisk-Users] QOS of VoIP
Hi All From wherewe can get the data for 1) ASR on various countries 2) Average Call drop on VoIP 3) Average Call Quality This we require to get an idea of what types of problem normally users use to face on voip and what is the average percentage of those problems. Pls. help me if anybody have the factsheet for various service provider on these paramaters Thanks Regards Ritesh Jalan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Errors Compiling chan_capi 0.3.5
On Mon, 30 May 2005, Mike Price wrote: On Thu, 2005-05-26 at 23:21, Armin Schindler wrote: On Thu, 26 May 2005, Mike Price wrote: Yes libcapi is installed. Here is a sample of the errors I am getting: In file included from chan_capi.c:38: chan_capi_pvt.h:92: syntax error before _cword chan_capi_pvt.h:92: warning: no semicolon at end of struct or union chan_capi_pvt.h:195: syntax error before '}' token chan_capi.c:41: syntax error before ast_capi_MessageNumber ... It looks like _cword is not defined. The definition is made in /usr/include/capiutils.h, do you have this file (should be installed by the libcapi/capi20 package) ? On what system do you compiling (SuSE, Debian, ... or even Cross) ? OK. I found I was missing isdn4k-utils-devel-3.1-76 and therefore capiutils.h etc. Now when I try to complie I get: chan_capi.c:114: variable `capi_tech' has initializer but incomplete type chan_capi.c:115: unknown field `type' specified in initializer chan_capi.c:115: warning: excess elements in struct initializer chan_capi.c:115: warning: (near initialization for `capi_tech') So things are still not quite right. The system is [EMAIL PROTECTED] v1.0 (CentOS 3.4 final) Any other pointers? It seems the setting in the Makefile of chan_capi about using CVS_HEAD is not correct. I don't know which version [EMAIL PROTECTED] v1.0 is using, but try to change that setting. Anyway, the CVS_HEAD/UNSTABLE_CVS defines in chan_capi is not really good. I changed that to have a small configure script, which makes the correct settings depending the Asterisk version it is compiled against. I will soon release my version of a reworked chan_capi. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for UK
Hmmmyes but last time I played with my FXO module on the TDM400 could not detect hangup properly (that is on a London BT line). Has this been fixed? I keep an eye on the CVS but I have not seen any fixes for that. Maybe I missed it. Vassilis Well, the official line is as Mr. Spencer has made in that bugtracker entry... Digium sell the TDM400P which supports polarity detection. CVS supports UK CallerID on that card. Digium no longer sell the X100P so it's not supported any more. The X100P is a fairly crappy choice for the UK since it has a hardcoded 600ohm impedance, suitable really only for the USA... But yes, 'it was only £10 on eBay' been there done that, wasted hours playing with txgain/rxgain/echo cancellation... :) Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Errors Compiling chan_capi 0.3.5
On Mon, 2005-05-30 at 18:21, Armin Schindler wrote: On Mon, 30 May 2005, Mike Price wrote: On Thu, 2005-05-26 at 23:21, Armin Schindler wrote: On Thu, 26 May 2005, Mike Price wrote: Yes libcapi is installed. Here is a sample of the errors I am getting: In file included from chan_capi.c:38: chan_capi_pvt.h:92: syntax error before _cword chan_capi_pvt.h:92: warning: no semicolon at end of struct or union chan_capi_pvt.h:195: syntax error before '}' token chan_capi.c:41: syntax error before ast_capi_MessageNumber ... It looks like _cword is not defined. The definition is made in /usr/include/capiutils.h, do you have this file (should be installed by the libcapi/capi20 package) ? On what system do you compiling (SuSE, Debian, ... or even Cross) ? OK. I found I was missing isdn4k-utils-devel-3.1-76 and therefore capiutils.h etc. Now when I try to complie I get: chan_capi.c:114: variable `capi_tech' has initializer but incomplete type chan_capi.c:115: unknown field `type' specified in initializer chan_capi.c:115: warning: excess elements in struct initializer chan_capi.c:115: warning: (near initialization for `capi_tech') So things are still not quite right. The system is [EMAIL PROTECTED] v1.0 (CentOS 3.4 final) Any other pointers? It seems the setting in the Makefile of chan_capi about using CVS_HEAD is not correct. I don't know which version [EMAIL PROTECTED] v1.0 is using, but try to change that setting. Anyway, the CVS_HEAD/UNSTABLE_CVS defines in chan_capi is not really good. I changed that to have a small configure script, which makes the correct settings depending the Asterisk version it is compiled against. I will soon release my version of a reworked chan_capi. I have changed the setting but now get different errors: gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DCVS_HEAD -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:69: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:70: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:71: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:72: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:73: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:74: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:75: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:114: variable `capi_tech' has initializer but incomplete type chan_capi.c:115: unknown field `type' specified in initializer chan_capi.c:115: warning: excess elements in struct initializer The version of Asterisk appears to be 1.0.7. When do you expect to have your reworked chan_capi ? Mike Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pre paid Card
Astcc works fine with me --- Rodrigo Otavio de Fraga [EMAIL PROTECTED] wrote: Hi, I liked to have a pre paid card in my asterisk Server. I saw some applications in the voip-ifo site, but all are not complete. Somebody has some tested and functioning solution ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BN8S0 problems (was: chan_misdn problem)
Ok, I have solved my problems by upgrading my chan_misdn and downgrading my mISDNuser. Now I have asterisk working with mISDN support. My problem now is that no matter what I do always see the link down. I've plugged the BN8S0 adapter to get the 8 ports working. When I plug to the ISDN box (using the Beronet crossing map recomendations) I have no response. I've tried to restart the port, restart the asterisk, even reload the modules, but always the link is down (on l1 and l2). Any hint?? Thanks. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Errors Compiling chan_capi 0.3.5
On Mon, 30 May 2005, Mike Price wrote: On Mon, 2005-05-30 at 18:21, Armin Schindler wrote: On Mon, 30 May 2005, Mike Price wrote: On Thu, 2005-05-26 at 23:21, Armin Schindler wrote: On Thu, 26 May 2005, Mike Price wrote: Yes libcapi is installed. Here is a sample of the errors I am getting: In file included from chan_capi.c:38: chan_capi_pvt.h:92: syntax error before _cword chan_capi_pvt.h:92: warning: no semicolon at end of struct or union chan_capi_pvt.h:195: syntax error before '}' token chan_capi.c:41: syntax error before ast_capi_MessageNumber ... It looks like _cword is not defined. The definition is made in /usr/include/capiutils.h, do you have this file (should be installed by the libcapi/capi20 package) ? On what system do you compiling (SuSE, Debian, ... or even Cross) ? OK. I found I was missing isdn4k-utils-devel-3.1-76 and therefore capiutils.h etc. Now when I try to complie I get: chan_capi.c:114: variable `capi_tech' has initializer but incomplete type chan_capi.c:115: unknown field `type' specified in initializer chan_capi.c:115: warning: excess elements in struct initializer chan_capi.c:115: warning: (near initialization for `capi_tech') So things are still not quite right. The system is [EMAIL PROTECTED] v1.0 (CentOS 3.4 final) Any other pointers? It seems the setting in the Makefile of chan_capi about using CVS_HEAD is not correct. I don't know which version [EMAIL PROTECTED] v1.0 is using, but try to change that setting. Anyway, the CVS_HEAD/UNSTABLE_CVS defines in chan_capi is not really good. I changed that to have a small configure script, which makes the correct settings depending the Asterisk version it is compiled against. I will soon release my version of a reworked chan_capi. I have changed the setting but now get different errors: gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DCVS_HEAD -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:69: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:70: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:71: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:72: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:73: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:74: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:75: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:114: variable `capi_tech' has initializer but incomplete type chan_capi.c:115: unknown field `type' specified in initializer chan_capi.c:115: warning: excess elements in struct initializer The version of Asterisk appears to be 1.0.7. Did you apply some patch to chan_capi? That's what I meant with the 'CVS_HEAD setting is not good', because some changes were made between some releases and the chan_capi Makefile just knows 'old' and 'new' which is not working for some versions. Anyway 1.0.7 should work with unpatched chan_capi 0.3.5. When do you expect to have your reworked chan_capi ? I still want to fix some race-conditions. It should be ready for 'testing' end of this week. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wacko Distinctive Ring Patterns being detected??
On Mon, May 30, 2005 at 09:03:29AM +1000, Gonzalo Servat wrote: On 5/29/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, May 28, 2005 at 06:34:23AM +1000, Gonzalo Servat wrote: [snip] If Asterisk allowed me to configure up to 10 ringing patterns, I could probably cover most of the ringing patterns being detected, but unfortunately there is a limit of 3 which means 50% (or more) of the calls are coming in under a distinctive ring pattern not configured in Asterisk, and hence going to the default context. Is there any deeper reason for that limitation, other than it didn't bother anybody enough? I wonder that myself, but I have no idea why the limit is imposed. Any Asterisk developers willing to answer that for us? Does anyone have any suggestions/ideas/etc on how to resolve this issue? Could you post here some ring patterns you get? A distinctive ring can identify a pattern that is similar enough to an existing pattern. You're right, some that were not defined were close enough to the ringing pattern and did match, but even with 3 popular distinctive rings defined there were still calls that were coming up with a new distinctive ring pattern and not getting matched by the defined dring patterns. Some of the ones I frequently saw were: 334,147,0 383,195,0 334,0,0 336,348,0 334,146,0 334,147,0 and 334,146,0 are practically the same. As for 334,0,0: Maybethe second patter was missed? I have the same problem here. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail make crash
Hi, I use asterisk-1.0.3 and mysql 4 on redhat 9.0. Normal call is OK, but when I use voicemail, * crash.Any one can help me?When voicemail app runs, * crash and no more trace. -- Executing VoiceMail("SIP/222.44.32.92-08155eb8", "u51292029") in new stackvmfax1*CLI> Disconnected from Asterisk serverUse GDB I found,gdb asterisk core.XX#0 0x415d312b in find_user (ivm=0x41c93e0c, context=0x0,mailbox=0x41c9425c "51292029") at mysql-vm-routines.h:7575 if(!strcmp(fields[i].name, "password")) {(gdb) bt#0 0x415d312b in find_user (ivm=0x41c93e0c, context=0x0,mailbox=0x41c9425c "51292029") at mysql-vm-routines.h:75#1 0x415d4923 in leave_voicemail (chan=0x8166168, ext=0x0, silent=0,busy=0, unavail=1) at app_voicemail.c:1332#2 0x415d9636 in vm_exec (chan=0x8166168, data="" at app_voicemail.c:3761#3 0x08073506
in pbx_exec (c=0x8166168, app=0x810ddb8,data="" newstack=1) at pbx.c:469#4 0x0807b3ce in pbx_extension_helper (c=0x8166168, context=0x81662c0"from-sip", exten=0x0, priority=1,callerid=0x80ffbc0 "\"856\" <856>", action="" at pbx.c:1277#5 0x0807551a in ast_pbx_run (c=0x41604c24) at pbx.c:1758#6 0x0807ba91 in pbx_thread (data="" at pbx.c:1981#7 0x4002f9b1 in pthread_start_thread () from /lib/i686/libpthread.so.0
??VIP???
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Re: [Asterisk-Users] Errors Compiling chan_capi 0.3.5
On Mon, 2005-05-30 at 19:27, Armin Schindler wrote: On Mon, 30 May 2005, Mike Price wrote: On Mon, 2005-05-30 at 18:21, Armin Schindler wrote: On Mon, 30 May 2005, Mike Price wrote: On Thu, 2005-05-26 at 23:21, Armin Schindler wrote: On Thu, 26 May 2005, Mike Price wrote: Yes libcapi is installed. Here is a sample of the errors I am getting: In file included from chan_capi.c:38: chan_capi_pvt.h:92: syntax error before _cword chan_capi_pvt.h:92: warning: no semicolon at end of struct or union chan_capi_pvt.h:195: syntax error before '}' token chan_capi.c:41: syntax error before ast_capi_MessageNumber ... It looks like _cword is not defined. The definition is made in /usr/include/capiutils.h, do you have this file (should be installed by the libcapi/capi20 package) ? On what system do you compiling (SuSE, Debian, ... or even Cross) ? OK. I found I was missing isdn4k-utils-devel-3.1-76 and therefore capiutils.h etc. Now when I try to complie I get: chan_capi.c:114: variable `capi_tech' has initializer but incomplete type chan_capi.c:115: unknown field `type' specified in initializer chan_capi.c:115: warning: excess elements in struct initializer chan_capi.c:115: warning: (near initialization for `capi_tech') So things are still not quite right. The system is [EMAIL PROTECTED] v1.0 (CentOS 3.4 final) Any other pointers? It seems the setting in the Makefile of chan_capi about using CVS_HEAD is not correct. I don't know which version [EMAIL PROTECTED] v1.0 is using, but try to change that setting. Anyway, the CVS_HEAD/UNSTABLE_CVS defines in chan_capi is not really good. I changed that to have a small configure script, which makes the correct settings depending the Asterisk version it is compiled against. I will soon release my version of a reworked chan_capi. I have changed the setting but now get different errors: gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DCVS_HEAD -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:69: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:70: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:71: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:72: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:73: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:74: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:75: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:114: variable `capi_tech' has initializer but incomplete type chan_capi.c:115: unknown field `type' specified in initializer chan_capi.c:115: warning: excess elements in struct initializer The version of Asterisk appears to be 1.0.7. Did you apply some patch to chan_capi? That's what I meant with the 'CVS_HEAD setting is not good', because some changes were made between some releases and the chan_capi Makefile just knows 'old' and 'new' which is not working for some versions. Anyway 1.0.7 should work with unpatched chan_capi 0.3.5. Thanks. I have now re-installed the unpatched chan_capi and have successfully compiled it. Now its on to try to get it to work. Thanks again When do you expect to have your reworked chan_capi ? I still want to fix some race-conditions. It should be ready for 'testing' end of this week. Let me know so I can give it a go. Mike Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk integration with Quintum Tenor AXT800!
Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio? i have a scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for intranet no PSTN at all just only IPphones connected through ehternet port and analog phones connected on FXS port.Is it's neccassary to connect with PSTN i don't want PSTN DIALING i'll just internal dialing.the scenerio is ipphones connected through ethernet while analog phones directly connected through FXS port is that possible i integrate Tenor AXT 800 in such a way that i describe above or may be i am asking a blind n dumb question Thw model number of voip gateway is Quintum Tenor AXT 800 with 8FXO,8FXS and 10/100Mbs LAN port. Kindly comments on that whether is that possible or not or what is the best way to utilize the power of Tenor gateway,practical experience working implementationc etc. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk@home
Can any one tell me what the mysql password, no its not password. [EMAIL PROTECTED] root]# mysql --user=root -p Enter password: Thx Q ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP clients
Hi, I have my asterisk server inside a NAT. When i connect a softphone SIP inside my net, all go well. Ok. But when I try connect a SIP softphone client outside my NAT, I get: NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex sip:[EMAIL PROTECTED]' failed for '83.41.119.25' Can someone help me with this? PD: Sorry for my english ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with SIP clients
Are you doing port forwarding on your firewall? Just make sure your asterisk port is open... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Piqueras Sent: 30 May 2005 10:49 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with SIP clients Hi, I have my asterisk server inside a NAT. When i connect a softphone SIP inside my net, all go well. Ok. But when I try connect a SIP softphone client outside my NAT, I get: NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex sip:[EMAIL PROTECTED]' failed for '83.41.119.25' Can someone help me with this? PD: Sorry for my english ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 registration period
Hi, Does anybody know how often does IAX2 registration happen ? Also I'm getting a feeling that there is no way of changing it through an iax.conf file ? Thanks, Ivan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to configure Inter7's Asterisk Fax with Postfix
I'm running Postfix as my email server. When comes to configuring Asterisk Fax, Inter7 recommends QMail. Does anybody knows postfix equivalent for the qmail? I don't have any knowlegde in QMail. This is the installation guide for astfax. ( http://www.inter7.com/astfax/INSTALL ) -- 3) Configure MTA [ This step is conditional upon your MTA software ] [ We, of course, recommend qmail ] mkdir /var/qmail/ast_fax cp ./ast_fax ./ast_fax.call /var/qmail/ast_fax vi /var/qmail/alias/.qmail-default: |/var/qmail/ast_fax/ast_fax /var/qmail/ast_fax/ast_fax.call chmod 600 /var/qmail/alias/.qmail-default -- Thanks. -eddie- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with SIP clients
Has you redirected all the RTP ports? You must redirect the SIP and the RTP streams. Take a look to the rtp.conf file of your asterisk installation to configure the RTP ports that you want to use. Best regards. Rpr Alex Piqueras escribió: Hi, I have my asterisk server inside a NAT. When i connect a softphone SIP inside my net, all go well. Ok. But when I try connect a SIP softphone client outside my NAT, I get: NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex sip:[EMAIL PROTECTED]' failed for '83.41.119.25' Can someone help me with this? PD: Sorry for my english ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Help
Preston Garrison wrote: Ever tried alot of sip devices on one asterisk box? You will see the need real fast :) Yes I know (I'm running SER with 400,000 user systems and Asterisk on the back end). However the OP was wanting it to solve a double NAT issue (when he has control of one of them), not for scale. :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with PrimuX 1S2M ISDN card
Hello, We have a PrimuX 1S2M ISDN card. This card has capi driver for linux-2.4. We are tying to configure it for Asterisk but we couldn't figure out how to do it. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_senddtmf.so.
Hi. I have installed asterisk 1.0.7. I need to send dtmf with tone duration. This functionality is in the cvs version. Can any body send me the app_senddtmf.so binary compiled for i386 or pentium IV to replace the 1.0.7 version.? I want to preserve the rest of 1.0.7 version of asterisk, because I use the debian package and I want to maintain the asterisk upgrades and patches. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_senddtmf.so.
On Mon, May 30, 2005 at 11:27:06AM +0200, Pepe Aracil wrote: Hi. I have installed asterisk 1.0.7. I need to send dtmf with tone duration. This functionality is in the cvs version. Can any body send me the app_senddtmf.so binary compiled for i386 or pentium IV to replace the 1.0.7 version.? I want to preserve the rest of 1.0.7 version of asterisk, because I use the debian package and I want to maintain the asterisk upgrades and patches. Are you aware of the existing debian package? http://packages.debian.org/unstable/comm/asterisk-app-dtmftotext http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=311008 And if you want a more up-to-date spandsp: http://tzafrir.org.il/rapid/ http://tzafrir.org.il/rapid/APT.html Just updates spandsp there yesterday and could use some testers :-) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
and , what is more interesting, they've omitted any reference to digium resellers and specified only distributors :( matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wacko Distinctive Ring Patterns being detected??
On Monday 30 May 2005 03:38, Tzafrir Cohen wrote: 334,147,0 and 334,146,0 are practically the same. As for 334,0,0: Maybethe second patter was missed? I have the same problem here. Yeah I was thinking the same thing; Does the distinctive ring code have any kind of filtering (say round each part of the cadence to the nearest 250ms)? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 to H323
Hi all, I'm using following software and equipment and I have very strange behavior: Asterisk CVS-NHEAD-05/30/05-16:42:41 H323 gatekeeper - GnuGK 2.2.2 IAX2 client - Firefly 1.9.8 build 3945 H323 client - SJPhone Build 1.50.271d H323 gateway - Welltech Wellgate 3504A When I dial from Firefly (IAX2) - SJPhone (H323) everything works as expected. When I dial from SJPhone (H323) - Firefly (IAX2) everything works as expected. When I dial from Firefly (IAX2) - Wellgate 3504A (H323) phone rings... I pick up... and from Firefly side ringing indication continues... no communication When I dial from Wellgate 3504A (H323) - Firefly (IAX2) Firefly rings... i pick up... from 3504A side i can hear the voice ... but Firefly continue ringing Does anyone experienced something similar. Any help will be appreciated. Here is dump from asterisk CLI with H323 debug and IAX2 debug on. Firefly to JSPhone (success) -- Executing Dial(IAX2/[EMAIL PROTECTED], H323/petew|30|t) in new stack -- Making call to petew using gatekeeper. == New H.323 Connection created. -- asterisk-1234 is calling host petew -- Call token is ip$localhost/13054 -- Call reference is 13054 -- DTMF Payload is [pt=101] -- Called petew Allowed Codecs: Table: G.711-uLaw-64k 1 UserInput/hookflash 2 UserInput/dtmf 3 Set: 0: 0: G.711-uLaw-64k 1 1: UserInput/hookflash 2 2: UserInput/dtmf 3 -- Sending SETUP message =-= In OnAlerting for call 13054: sessionId=101 -- Ringing phone for petew - Progress Indicator: 0 -- H323/petew-5 is ringing Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: RINGING Timestamp: 9ms SCall: 1 DCall: 04596 [192.168.0.70:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 9ms SCall: 04596 DCall: 1 [192.168.0.70:4569] -- H323/petew-5 is ringing -- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 1 External RTP Session Starting RTP channel id 1 parameters: -- Started logical channel: sending G.711-uLaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.0.70 -- remotePort: 49156 -- ExternalIpAddress: 192.168.8.1 -- ExternalPort: 13072 =-= In OnConnectionEstablished for call 13054 -- Connection Established with petew -- H323/petew-5 answered IAX2/[EMAIL PROTECTED] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 002 Type: CONTROL Subclass: (255?) Timestamp: 03446ms SCall: 1 DCall: 04596 [192.168.0.70:4569] Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 03449ms SCall: 1 DCall: 04596 [192.168.0.70:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 03446ms SCall: 04596 DCall: 1 [192.168.0.70:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 03449ms SCall: 04596 DCall: 1 [192.168.0.70:4569] Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 002 Type: VOICE Subclass: 4 Timestamp: 03460ms SCall: 1 DCall: 04596 [192.168.0.70:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 005 Type: VOICE Subclass: 4 Timestamp: 03474ms SCall: 04596 DCall: 1 [192.168.0.70:4569] Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 03474ms SCall: 1 DCall: 04596 [192.168.0.70:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 03460ms SCall: 04596 DCall: 1 [192.168.0.70:4569] Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10001ms SCall: 1 DCall: 04596 [192.168.0.70:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 007 Type: IAX Subclass: LAGRP Timestamp: 10001ms SCall: 04596 DCall: 1 [192.168.0.70:4569] Tx-Frame Retry[-01] -- OSeqno: 007 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10001ms SCall: 1 DCall: 04596 [192.168.0.70:4569] Firefly to 3504A (failure) -- Executing Dial(IAX2/[EMAIL PROTECTED], H323/20001|30|t) in new stack -- Making call to 20001 using gatekeeper. == New H.323 Connection created. -- asterisk-1234 is calling host 20001 -- Call token is ip$localhost/13055 -- Call reference is 13055 -- DTMF Payload is [pt=101] -- Called 20001 Allowed Codecs: Table: G.711-uLaw-64k 1 UserInput/hookflash 2 UserInput/dtmf 3 Set: 0: 0: G.711-uLaw-64k 1 1: UserInput/hookflash 2 2: UserInput/dtmf 3 -- Sending SETUP message -- Started logical channel:
[Asterisk-Users] @home to @home
Hi Is there a way that you can setup 2 [EMAIL PROTECTED] boxes, to communicate with each other. Example: Caller 1 [EMAIL PROTECTED] Internet [EMAIL PROTECTED] Caller 2 Thx Q ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960s and skinny
Hi Andy, Thank you so much for your help. My 7910's are now working!!!:) Now I can work with those IP phones. I am still monitoring them. I will let you know the further status. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Serious ZapRAS problem!
Hi! I've been trying to get ZapRAS or PPPD to work. Neither does! All i get is LCP: timeout sending Config-Requests But after trying, all voicelines get crazy! It sounds like robots when somebody calls! And since the zaptel drivers can't unload (the server hangs totaly if I try!), I have to reboot the whole server! The robot-voice is only on our side, it sounds fine at the other end. My server is an Dell PowerEdge 1850 with an Digium TE110P card to PRI. I'm using stable Asterisk 1.0.6 and I'm located in Sweden with TDC Song as telco. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: HiPath 4000 and Asterisk
Hi,Could you please post your oh323.conf file and explain which changes are required at the HiPath? ( I have the HG3550 Card ) however I have no access to the HiPath system and I need to ask someone else to perform that changes therefore my ability to debug this issue is small.Thanks a lot!Ohad--- Ohad.Levy at infineon.com wrote: I'm trying to setup Asterisk trunk to Siemens HiPath 4000 V2.01i suppose you mean version 2.0 ;-) What would be the best way to do so? I am a bit confused because as far as I've understand this PBX doesn't support H323, but I saw somewhere someone who created a cornet trunk and it worked using H323.I have a HiPath4000 V1.0 interconnected to Asteriskusing a STMI board (HG3550) and oh323. Theinteroperability works well. The chan_cornet AFAIK isnot released by Steffen. The interconnection betweenH4kV2.0 and * is identical, use a HG3550 V2.0 for theH4k and oh323 for *. I've read some information about the cornet connectivity which is in development - does anyone knows the status of that?AFAIK not released :-( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX encrytion
What encryption features are available to encrypt the IAX2 traffic between two asterisk servers. I have read that there is some encryption possible but has anyone been able to encrypt the entire payload of IAX traffic between two asterisk servers. regards, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
Matteo Brancaleoni wrote: and , what is more interesting, they've omitted any reference to digium resellers and specified only distributors :( Yah that's kinda bad form. And seeing as I know for a fact that the distributors will sell directly to my customers even though we're in different countries... -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] IAX encrytion
IpSec VPN ;-) Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de John Melody Envoyé : lundi 30 mai 2005 10:37 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] IAX encrytion What encryption features are available to encrypt the IAX2 traffic between two asterisk servers. I have read that there is some encryption possible but has anyone been able to encrypt the entire payload of IAX traffic between two asterisk servers. regards, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP and missing digit map
Hi all, I'm trying to setup an MGCP connection between my asterisk and a third party pbx system. I have very little control over the external pbx. The calls are failing with the following asterisk error: notice chan_mgcp.c 2347 handle_repsonse: Terminating on result 519 from aaln/[EMAIL PROTECTED] Searching the net shows that the result 519 relates to a digit map. It seems the pbx is requesting a digit map from Asterisk and failing because it is not getting one. Does asterisk/mgcp support this digit map feature? If it does what are the config settings? If asterisk does not support this feature what changes should I be asking the pbx support person to make to his system? I'm running fedora core 1 and Asterisk CVS-D2004.12.21.13.00 thanks in advance bruce ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AT-320 + supervised transfer
Hi all, I'm trying attended transfer on Asterisk 1.0.7 and AT-320 phone. I met a lot of problems during this steps, while in the blind transfer all works fine. I had this kind of problem: CASE 1: A call B B seton hold A B call C (that is busy for some reason) B try to get the first call with "hook flash" (or pressing the "hold" key)and A stop to work. B stay waiting with an hung call with C without have the possibility to re-take it CASE 2: A call B B set pn hold A B call C (that is busy for some reason) B is talking with C B has need to speak again with A, so press the "hold" key. At this point A and C is frozen and B stay waiting with an hung call lost somewhere Another problem that i found, is that just i switch on the phone, i can't make and receiveany call until (the phone seem too slow to boot). I have to "pick up" and"pickdown" the handset I contattedthe chip's productor and he said me that my server (Asterisk) doesn't have the re-invite. The only thingh that i can do is to set "canreinvite = yes" in my sip.conf Anyone can help me please? Anyone never had a problem similar to mine ? Thanks for all Giordano ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pls, find me a VoIP Supplier/Reseller in Dubai-UAE
Hi, All I am looking for Suppliers/resellers from Dubai - UAE to buy some VoIP products and Digium's TDM cards. could some one send me some contact information in this regards?. mainly I want to buy Hardware SIP phones, VoIP gateways and DTM cards (FXS). Thank you Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash Operator Panel 0.21 released
After a while, version 0.21 of the Flash Operator Panel is out. Flash Operator Panel displays information about your Asterisk PBX activity in real time via a standard web browser with Flash plugin. It can monitor several asterisk servers at once. It can integrate with CRM software, by poping up a web page (and passing the CLID and CLIDNAME) when a specified button is ringing. It also can be used to enable click-to-dial for web based applications. It can monitor almost any asterisk channel type available: ZAP, SIP, IAX2, H323, OH323, MGCP, CAPI, MODEM/I4L, VPB, mISDN, etc. Monitoring Features: -Monitoring of Agents (logged in/off) -Monitoring of Queues -Monitoring of Parked calls/slots -Monitoring of conferences -Shows ip address of sip/iax2 peers -Shows sip/iax2 status/reachability -Shows callerid/called number -Shows timers/countdown for absolute timeout calls or parked slots -Shows statistics on agents and queues Available Actions: -Hangup a channel -transfer via dragdrop -originate via dragdrop -Set absolute timeout when transferring -Set callerid when transferring -Reload Asterisk -Mute/Unmute meetme participants -Barge-in on a call (optionally barge-in muted to avoid being noticed) -Commands optionally restricted by security code -Commands optionally restricted to a specific button/channel The new version has lots of new features, such as: *REGEXP buttons (they replace wildcard buttons) *Set timeout for transferred calls *Change state/label/text for buttons based on astdb values or dialplan userevents *Fire popups from the dialplan passing any channel variable to the web application The almost complete list of changes can be seen here: http://www.asternic.org/CHANGES.html You can download it, read documentation, browse the mailing list archives or subscribe to it from: http://www.asternic.org You can also rate the project on freshmeat: http://freshmeat.net/rate/53045/ Have a good day! -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN RAS and data calls
It seems like when I use PPPD-command, or ZapRAS, Asterisk doesn't make it a data call, but a regular voice-call. My ISP change their behaivour depending on the incoming call-type (data or voice). If it's voice, they try to open up a V.90 connection. Else (data call) it will reply with PPP directly. The both methods uses the same modem pool number. How can I tell Asterisk to inititate a data-call instead of voice-call? I'm using .call-files to connect to the ISP. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Soekris
I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Areski Calling Card
Does anyone have complete Areski Calling Card directions? I think Areski Calling Card Idiots guide is not complete :( any other guide? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Digium Website Update: Asterisk Business Edition
* Andrew Kohlsmith [EMAIL PROTECTED] [050529 21:07]: On Sunday 29 May 2005 20:59, Aidan Van Dyk wrote: 1) Simply CVS head (as of some point in time) with certain features or bug fixes backed out 2) In addition to CVS head, some important features and bug fixes. I think it's simply #2. They are taking HEAD and maintaining a version where they are extraordinarily careful about what goes in. Similar to what stable was supposed to be. Well, I disagree on the stable comment here - but that's neither here nor there. If they are fixing things in ABE that are broken in Open Source, and not simply backing out features, they why aren't these bug fixes in the open source version? If they are only backing out features to be more conservative, then that would quite OK, but the truth is, they haven't really been up front with what they are doing. But part of the point of the GPL is that these things are open. But Digium doesn't have to comply with the GPL for stuff properly disclaimed, so they are allowed to do that. I've just not been impressed with Digium's behaviour lately. They've gotten quite hostile over Sangoma hardware lately, claiming that Sangoma (by continuing to develop, refine, and expand their hardware lines, which are much older than asterisk, and which asterisk was originally developed on) are just ripping them off. If anything, Digium is ripping people off with hardware which is inferior (though I've seen claims that they have some good new stuff coming - excellent!). And now they introduce a product that *directly* competes with the people in the asterisk community (is that Digium speak for free coders?) who are working on developing, using, and selling asterisk-based solutions. What incentive does some non-Digium person/developer/company now have to make sure their stuff is disclaimed to Digium, if they know that doing so will give Digium a leg up on them in trying to sell it? And for all the claims we've heard that there is no other version of asterisk, we know know that Digium does, indeed, have such a version, which, though based on the available open-source code, is different, and that the differences are unknown, and that they are selling it as direct competition to the others contributing to Asterisk. In all, we've seen Digium going to great lengths to try and build an asterisk community to enhance asterisk, but is treating them as a group of indentured users, testers, coders. a. -- Aidan Van Dyk Create like a god, [EMAIL PROTECTED] command like a king, http://www.highrise.ca/ work like a slave. signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gradwell UK DID + DTMF
Does anyone have a Gradwell UK SIP number successfully receiving DTMF working with their Asterisk? If so, please could you post the relevant bits of your config files. Thanks in advance Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENTt: GPL Asterisk Billing Software
Partially. I have not finished the script that will limit the calls depending on the money available. Darren Wiebe [EMAIL PROTECTED] VoIP Newbie wrote: Does it support pre-paid billing? On 5/30/05, Darren Wiebe [EMAIL PROTECTED] wrote: El Flynn wrote: Darren Wiebe wrote: Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. The link in the original email opens a page that says Download the latest version of the code from http://www.aleph-com.net/astpp.html Has anyone else been able to download this code? I can't seem to find a link on their site to the code itself, and the astpp.html page brings up a Not Found... Sorry, I missed that old link. I just got everything moved onto the wiki on Friday night. Please download the code off of the cvs server. I'm getting close to ready to release version 1.0 and then I will post a copy on the website. At present, I believe the only show stopping bug is in the AgileBill integration. That will be fixed shortly. Darren Wiebe [EMAIL PROTECTED] Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 60 second time out
On Sun, 2005-05-29 at 14:41 -0400, C F wrote: This is not the CLI output. Please reproduce the problem and paste the CLI output, from both, when it's set to 10 seconds, and when It's set to 60. Did you remember to answer the call before passing to voicemail?? Some PSTN providers will drop the call after 60secs of ringing with no answer. You might also want to decrease the call time to 19 seconds per person to try and avoid the 60 second timeout... See below... Regards, Adam exten = 2001,1,Dial(sip/7780,19) exten = 2001,2,Goto(2001,102) exten = 2001,102,Dial(sip/7781,19) exten = 2001,103,Goto(2001,203) exten = 2001,203,Dial(sip/7782,19) exten = 2001,204,Goto(2001,304) exten = 2001,304,Answer exten = 2001,305,VoiceMail2(u7782) exten = 2001,306,Hangup -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Serious ZapRAS problem!
On Mon, 2005-05-30 at 13:26 +0200, Daniel Nystrm wrote: My server is an Dell PowerEdge 1850 with an Digium TE110P card to PRI. I'm using stable Asterisk 1.0.6 and I'm located in Sweden with TDC Song as telco. Try upgrading to current stable, which is either 1.0.7 or else CVS - STABLE. Also, as per your other email, see if your provider can assign a separate dedicated ISDN DID for you Other than those two suggestions, I can't help, sorry. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENTt: GPL Asterisk Billing Software
What happens if the rate changes mid call? IE, call starts @ 18.30 and lasts till 19.15 Rate changes @1900 to off-peak. Darren Wiebe wrote: Partially. I have not finished the script that will limit the calls depending on the money available. Darren Wiebe [EMAIL PROTECTED] VoIP Newbie wrote: Does it support pre-paid billing? On 5/30/05, Darren Wiebe [EMAIL PROTECTED] wrote: El Flynn wrote: Darren Wiebe wrote: Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. The link in the original email opens a page that says Download the latest version of the code from http://www.aleph-com.net/astpp.html Has anyone else been able to download this code? I can't seem to find a link on their site to the code itself, and the astpp.html page brings up a Not Found... Sorry, I missed that old link. I just got everything moved onto the wiki on Friday night. Please download the code off of the cvs server. I'm getting close to ready to release version 1.0 and then I will post a copy on the website. At present, I believe the only show stopping bug is in the AgileBill integration. That will be fixed shortly. Darren Wiebe [EMAIL PROTECTED] Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CRM integration (was RE: CallerID)
On Sat, 2005-05-28 at 19:19 +0100, Tom Fanning wrote: snip The guy mentioned Java from within the browser. I believe that I am right in saying that a Java applet should very well be able to listen for tcp connections as well as udp datagrams. Try this primer: http://homepages.uel.ac.uk/2795l/pages/javaapps.htm#Class%20ServerSocket%20( TCP%20Server%20Connections) Yep, thanks for replying for me... So, has anyone got the time + motivation to do something??? I wish I did :( Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AT-320 + supervised transfer
Hi, 1.0.7 does not support atxfer. You nees to use CVS. I have a bunch of the ATCOM cheapos and only CVS will take notice of 'atxfer' in features.conf. Otherwise , consider this scenario... Call comes in, press HOLD, dial other party to see if they wish to speak to the caller. If so, press * to hang up, then HOLD to swap back to the incoming caller. Announce you are going to transfer them, and now dial the final extension and press FWD to do a blind transfer. This works for me with the SIP 1.43 firmware. The IAX fw still has some way to go... The phone seems slow to boot? Ensure you have the IP/hostname of a valid NTP time server at the bottom of the web config page. If you don't, it will take ages and eventually fall back on 'time.windows.com' Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Soekris
See: http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=3 damn cool. -Original Message- From: Chris Mason (Lists) [mailto:[EMAIL PROTECTED] Sent: Monday, May 30, 2005 7:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk on Soekris I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
Chris Mason (Lists) wrote: I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If I'm not mistaken, the Soekris hardware does fine for a few voice channels - but not a very high performance piece of hardware. For example, if you wanted a full solution as a VPN, Asterisk server, media streaming via ICEs, web server, email server, etc... it will start to lack in performace when compared to a VIA EDEN system which can use DDR memory and such. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voice is coming only from one side
trye ILBC codec, and make sure that the phone configuration will use ILBC, I dont mean the sip.conf configuration, but the specific phone client configuration that some phones allow. Alsol try to use ILBC in sip.conf putting disallow=all, allow=ilbc give that a try and please post here your sip.conf and the output that asterisk shows in the console being verbose (asterisk -vvvr) best regards On 5/29/05, Nil s [EMAIL PROTECTED] wrote: Hello, I am new asterisk user. I am trying to setup asterisk locally. I have installed Red Hat 9.0 on my PC and I installed asterisk on it. Then i configured sip.conf, Extensions.conf, voicemail.conf for two users. I am using Soft dialer to make calles. I have two another PC's. So all the three PC's including Astersik server are in Local LAN. I configured Softphones for those two users on two PC's. I tried to call from one number to another, call is correctly established however voice is not coming at one end. Means i can listen other users voice but he cant. I have tried to allow different codecs in sip.conf such as gsm, g711 etc. But No luck. Please help me. One more thing is that voicemail is perfectly working for both the users. Only problem is that Other user can not hear me. Thanks, Nil. Do You Yahoo!? Yahoo! Small Business - Try our new Resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
Chris Mason (Lists) wrote: I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason Chris, You sure did see a Soekris board! They actually can run Asterisk quite well, but you will want to check out my project (AstLinux) for more information: http://www.kriscompanies.com/modules.php?name=Contentpa=showpagepid=3 -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk compatible, hot swappable PRI card
Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remove PRI cards without shutting down the system Is there such a thing as an asterisk compatible hot-swappable PRI card and board ? Someone told me to look at the C-PCI technology, it seems that telecom company use this. Thanks Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] choice of processors
Hi there I am moving into a production environment. I will mostly be using Meetme, with Ztdummy for timing. I have a question on which of 2 processor setups is favourable. I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4 3.06GHz Processor. These will cost me exactly the same amount. Would one of these processor setups be favourable, both in terms of performance and running Asterisk? Many thanks Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
Dustin Wildes wrote: Chris Mason (Lists) wrote: I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If I'm not mistaken, the Soekris hardware does fine for a few voice channels - but not a very high performance piece of hardware. For example, if you wanted a full solution as a VPN, Asterisk server, media streaming via ICEs, web server, email server, etc... it will start to lack in performace when compared to a VIA EDEN system which can use DDR memory and such. Of course it would start to lack in performance! You'd have to be CRAZY to run all of that on a fanless $220 SBC! Like anything else, the Soekris is not an end-all, be-all solution. It does however, work surprisingly well in a lot of different applications and I am routinely impressed when I hear what people are doing with them (and AstLinux). :) -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] choice of processors
The Dual 2.8GHz will be much faster for running everything. If it is the same price it should be a no brainier, take the two CPU system. Depending on the manufacture of the system it may even take a failure of one CPU. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Langley Sent: May 30, 2005 11:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] choice of processors Hi there I am moving into a production environment. I will mostly be using Meetme, with Ztdummy for timing. I have a question on which of 2 processor setups is favourable. I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4 3.06GHz Processor. These will cost me exactly the same amount. Would one of these processor setups be favourable, both in terms of performance and running Asterisk? Many thanks Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] choice of processors
Need to provide a little more info: What's the bus speed? What kind of motherboard would you use with each? What kind of RAM at what speed? What cache size are on the CPUs? Also, what price are these as equals? I've seen two Xeon 2.8GHz 800MHz processors for about US$450 and a single P4 at the same price would be a 3.6GHz 2MB cache at several resellers. which resellers do you buy through or are you buying a prebuilt system? MATT--- -Original Message- From: Steven Langley [mailto:[EMAIL PROTECTED] Sent: Monday, May 30, 2005 11:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] choice of processors Hi there I am moving into a production environment. I will mostly be using Meetme, with Ztdummy for timing. I have a question on which of 2 processor setups is favourable. I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4 3.06GHz Processor. These will cost me exactly the same amount. Would one of these processor setups be favourable, both in terms of performance and running Asterisk? Many thanks Steven ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
Maybe my point was missed. Hardware wise - a VIA MII EDEN based board will greatly outperform a Soekris system, which is why my embedded platform is based on the VIA hardware instead of the Soekris, because I AND my customers did want an all-in-one system, and small offices tend to want an all-in-one piece of equipment. Kristian Kielhofner wrote: Dustin Wildes wrote: Chris Mason (Lists) wrote: I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If I'm not mistaken, the Soekris hardware does fine for a few voice channels - but not a very high performance piece of hardware. For example, if you wanted a full solution as a VPN, Asterisk server, media streaming via ICEs, web server, email server, etc... it will start to lack in performace when compared to a VIA EDEN system which can use DDR memory and such. Of course it would start to lack in performance! You'd have to be CRAZY to run all of that on a fanless $220 SBC! Like anything else, the Soekris is not an end-all, be-all solution. It does however, work surprisingly well in a lot of different applications and I am routinely impressed when I hear what people are doing with them (and AstLinux). :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] AT-320 + supervised transfer
Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to emprove atxfer...how asterisk emprove the atxfer ? :| How do u set your sip.conf for the at-320 ? Did u set the canreinvite option ? Thanks for all, Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: lunedì 30 maggio 2005 16.22 A: asterisk-users@lists.digium.com Oggetto: RE: [Asterisk-Users] AT-320 + supervised transfer Hi, 1.0.7 does not support atxfer. You nees to use CVS. I have a bunch of the ATCOM cheapos and only CVS will take notice of 'atxfer' in features.conf. Otherwise , consider this scenario... Call comes in, press HOLD, dial other party to see if they wish to speak to the caller. If so, press * to hang up, then HOLD to swap back to the incoming caller. Announce you are going to transfer them, and now dial the final extension and press FWD to do a blind transfer. This works for me with the SIP 1.43 firmware. The IAX fw still has some way to go... The phone seems slow to boot? Ensure you have the IP/hostname of a valid NTP time server at the bottom of the web config page. If you don't, it will take ages and eventually fall back on 'time.windows.com' Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX encrytion
I am using vtun without incident: http://vtun.sourceforge.net/ this is a bolt-on and depending on the box's specs and the number of tunnels, it may negatively impact the server's performance. To address this, in my application, I use a seperate box to aggregate all of the remote IAX servers tunnels and marshall all of the traffic to my primary server. The seperate box is a lowly P-II 400 and it works fine with 25 tunnels going into it. hth -Original Message- From: John Melody [mailto:[EMAIL PROTECTED] Sent: Monday, May 30, 2005 2:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] IAX encrytion What encryption features are available to encrypt the IAX2 traffic between two asterisk servers. I have read that there is some encryption possible but has anyone been able to encrypt the entire payload of IAX traffic between two asterisk servers. regards, John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
If I'm not mistaken, the Soekris hardware does fine for a few voice channels - but not a very high performance piece of hardware. For example, if you wanted a full solution as a VPN, Asterisk server, media streaming via ICEs, web server, email server, etc... it will start to lack in performace when compared to a VIA EDEN system which can use DDR memory and such. Of course it would start to lack in performance! You'd have to be CRAZY to run all of that on a fanless $220 SBC! Like anything else, the Soekris is not an end-all, be-all solution. It does however, work surprisingly well in a lot of different applications and I am routinely impressed when I hear what people are doing with them (and AstLinux). :) I'm starting a project in the near future, and am have seen the question raised on the list, but no answer yet.. so 2 questions... How many channels could this board deal with when purely translating from G.729 IAX2 - G.729 SIP ie, is there a codec translation included in that step, or is it optimised such that you just change the 'headers' ?? Could you do 10 or even 20 such calls ?? Basically the idea would be to use this box to convert those annoying but decent business phones from SIP to IAX2 with trunking If it works well, would probably also look at utilising it for QoS/DHCP and FTP (to config the phones) but that's about it Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
On Monday 30 May 2005 11:28, Adam Goryachev wrote: How many channels could this board deal with when purely translating from G.729 IAX2 - G.729 SIP That's not a codec translation; Asterisk can simply take the IAX2 audio frames and stuff them into RTP frames without actually deconstructing the audio itself and recoding it. As far as how many -- this is where you must do the research and post results. There is no data on it at this point, mostly because people just haven't done the research. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meridian 808 Function
Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx programation)? Thanks for the hint Miguel Ruiz Velasco Version: OpenKeyServer v1.2 Comment: Extracted from belgium.keyserver.net Signature: 0x59831109 __ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] AT-320 + supervised transfer
On Monday 30 May 2005 16:19, Giordano Grandis wrote: Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. Well, I run a local NTP server, so it's as fast plus has the correct time at the end :) I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to emprove atxfer...how asterisk emprove the atxfer ? :| When Asterisk does the transfer natively, the procedure is like this: Call comes in, hold on I'll try to transfer you. you dial *2 (or any sequence you define), speak to the remote party. If they want to speak to the caller, YOU hang up. If they don't, THEY hang up and you are returned to the original caller :) How do u set your sip.conf for the at-320 ? Did u set the canreinvite option ? [1300] type=friend username=1300 secret=ahem host=dynamic context=from-ip nat=yes canreinvite=no Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Help
Hi Matt,
Can I know your setting in asterisk to allow
connection from SER?
Below is my configuration in sip.conf : -
; incoming calls from ser
[ser-in]
type=friend
host=1.1.1.2
And, can I have your SER configuration file ser.cfg?
Below is my ser.cfg config: -
if (uri=~1.1.1.4) {
if (method==REGISTER) {
if (!www_authorize(1.1.1.4,
subscriber)) {
www_challenge(1.1.1.4, 0);
break;
};
save(location);
break;
};
if (!lookup(location)) {
forward( 1.1.1.3, 5060 );
break;
};
};
Regards,
rootlinux
--- Matt Riddell [EMAIL PROTECTED] wrote:
Preston Garrison wrote:
Ever tried alot of sip devices on one asterisk
box? You will see the
need real fast :)
Yes I know (I'm running SER with 400,000 user
systems and Asterisk on
the back end).
However the OP was wanting it to solve a double NAT
issue (when he has
control of one of them), not for scale.
:D
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Asterisk on Soekris
Andrew Kohlsmith wrote: On Monday 30 May 2005 11:28, Adam Goryachev wrote: How many channels could this board deal with when purely translating from G.729 IAX2 - G.729 SIP That's not a codec translation; Asterisk can simply take the IAX2 audio frames and stuff them into RTP frames without actually deconstructing the audio itself and recoding it. Exactly. I have people doing at least ~30 calls in that manner. That's just glorified packet forwarding (okay not really, but I can't think of anything better). As far as how many -- this is where you must do the research and post results. There is no data on it at this point, mostly because people just haven't done the research. :-) -A. I know that I keep using this as an excuse, but with Asterisk there are just too many variables to be able to answer questions like How many calls x x? Codecs, protocols, trunking, re-invites, echo can., etc, make it very difficult to come up with numbers. I want to do an astertest on this hardware, but have just not had enough time... -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meridian 808 Function
On Mon, 30 May 2005 08:53:12 -0700 (PDT), Miguel Ruiz Velasco Sobrino wrote Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx programation)? Thanks for the hint The way to use is (which I do not really recommend) is that the nortel user will dial the extension that is connected to the Asterisk server and then press the FUNCTION button on his/her phone followed by 808. After 1 or 2 seconds the phone will say Long Tones. You can then dial the extension you want on the * server. The best way to integrate with a Meridian is to use FXS ports connected to trunk line ports and then configure the PBX to dial something like 7 to get a dial tone from the Asterisk server. That way your users do not have to remember a complicated procedure. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
Dustin Wildes wrote: Maybe my point was missed. Hardware wise - a VIA MII EDEN based board will greatly outperform a Soekris system, which is why my embedded platform is based on the VIA hardware instead of the Soekris, because I AND my customers did want an all-in-one system, and small offices tend to want an all-in-one piece of equipment. Dustin, Yes, a VIA Eden board will greatly outperform a Net4801. My CL1 is actually quite powerful. However, several points on the mini-itx architecture that need to be mentioned: 1) Heat/Reliability. Much more heat generated, my mini-itx system has three fans. The Soekris has none (not even a heatsink). This makes the Soekris much more reliable - no moving parts. 2) Power usage. All though I have yet to measure it, my mini-itx system has a 90 watt power supply (which is ATX based, btw). My Soekris has a 12 watt power supply. Also on another note of reliability, I trust the Soekris power supply much more than the half breed ATX in most mini-itx systems. Yes, I do know that just because you have a 90 watt power supply you are not using all 90 watts, but the fact that the Soekris has a 12 watt power supply means that it is DEFINITELY not using more than 12 watts. 3) Cases. Have you been able to find a reasonably priced case for mini-itx that doesn't look like some cheap home theater appliance? I haven't. One thing often looked for (especially in the embedded space) is for the device to look like an appliance. People are much less likely to mess with something when they don't know what it is. With a mini-itx case with upfront firewire and line-out, my 14 year old cousin would have his fingers in that case in a minute! When the 7501 comes out later this year there won't even be a point of arguing this anymore. That board is going to be killer! Your point was not missed, but I don't think it is a good idea to include that much hodge podge functionality (web server, mail server, PBX, streaming media server, etc, etc) in one system. Also, most of my customers want reliability. Which the Soekris has over the ITX stuff, hands down. Also, as far as AstLinux is concerned, I don't care whether you are using a Soekris or not. That is why I have a generic i586 image! It works perfectly on the mini-itx boards, and in fact, is what my CL1 is running right now. P.S. - Please apply :) or ;) to this whole message. It might sound like I got pretty fired up, but that is almost never the case with me! -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: [Asterisk-Users] AT-320 + supervised transfer
The procedure that will do asterisk is very nice ;) but whe it was available ? Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. Here my sip.conf for the phone, can u say me if there is somethingh wrong ? [2391] type=friend username=2391 secret=2391 language=it host=dynamic context=intern dtmfmode=rfc2833 callgroup=1 pickupgroup=1 I think is ok, maybe i have some problem on phone settings.Can I see your exmple phone setting ? Thanks, Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: lunedì 30 maggio 2005 18.04 A: asterisk-users@lists.digium.com Oggetto: Re: R: [Asterisk-Users] AT-320 + supervised transfer On Monday 30 May 2005 16:19, Giordano Grandis wrote: Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. Well, I run a local NTP server, so it's as fast plus has the correct time at the end :) I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to emprove atxfer...how asterisk emprove the atxfer ? :| When Asterisk does the transfer natively, the procedure is like this: Call comes in, hold on I'll try to transfer you. you dial *2 (or any sequence you define), speak to the remote party. If they want to speak to the caller, YOU hang up. If they don't, THEY hang up and you are returned to the original caller :) How do u set your sip.conf for the at-320 ? Did u set the canreinvite option ? [1300] type=friend username=1300 secret=ahem host=dynamic context=from-ip nat=yes canreinvite=no Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for UK
On Monday 30 May 2005 07:22, Vassilis Konstantinou wrote: Hmmmyes but last time I played with my FXO module on the TDM400 could not detect hangup properly (that is on a London BT line). Has this been fixed? I keep an eye on the CVS but I have not seen any fixes for that. Maybe I missed it. No idea... I didn't say the TDM400P was actually any good (I don't own one), just that it is better supported than the X100P :) gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] nntp access
hi, is it possible to get to this group via nntp ? Regards, Marcin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] perl agi : get_variable problem
Hi, I'm developping some AGI in perl (5.8.6) on i386 using Asterisk 1.0.5. I want to get some variables such as DIALSTATUS and ANSWEREDTIME after a $AGI-exec(Dial, dial_string); but here is what i get actually: DIALSTATUS= DIALEDTIME=ANSWER ANSWEREDTIME=18 I searched the archives and saw that $AGI-verbose could mess the access to variables, but I don't use it. Any clue welcome :-) Julien -- # Key fingerprint = 121C BB26 F4EE 59AF 5E68 7020 5BC9 D7AB 4CBE C6AD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Monday 30 May 2005 17:22, Giordano Grandis wrote: The procedure that will do asterisk is very nice ;) but whe it was available ? Asterisk's atxfer support is only in CVS. Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. You *must* be using a new firmware for the phone. Download 1.43 from http://www.aredfox.com/edownloadssip.htm (the AT-320 needs PA186S code) Here my sip.conf for the phone, can u say me if there is somethingh wrong ? Looks fine to me.. I think is ok, maybe i have some problem on phone settings.Can I see your exmple phone setting ? They're at work so I can't see the config right now... but they're just the defaults with the DTMF changed to RFC2833 and the NTP server set... Try resetting to defaults using the procedure at http://www.voip-info.org/wiki-ATCOM+AT-320 Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
Not a problem Kristian! :-) Same here! Comments below: Kristian Kielhofner wrote: Dustin Wildes wrote: Maybe my point was missed. Hardware wise - a VIA MII EDEN based board will greatly outperform a Soekris system, which is why my embedded platform is based on the VIA hardware instead of the Soekris, because I AND my customers did want an all-in-one system, and small offices tend to want an all-in-one piece of equipment. Dustin, Yes, a VIA Eden board will greatly outperform a Net4801. My CL1 is actually quite powerful. However, several points on the mini-itx architecture that need to be mentioned: 1) Heat/Reliability. Much more heat generated, my mini-itx system has three fans. The Soekris has none (not even a heatsink). This makes the Soekris much more reliable - no moving parts. I am using the MII 6000 (no fans) with a heatpipe to replace the embedded heatsink - pushing to extruded fins. It does get warm, but not that bad. 2) Power usage. All though I have yet to measure it, my mini-itx system has a 90 watt power supply (which is ATX based, btw). My Soekris has a 12 watt power supply. Also on another note of reliability, I trust the Soekris power supply much more than the half breed ATX in most mini-itx systems. Yes, I do know that just because you have a 90 watt power supply you are not using all 90 watts, but the fact that the Soekris has a 12 watt power supply means that it is DEFINITELY not using more than 12 watts. I haven't measured the power either - but we have been using the morex power supplies for several months now, and no problems. But I not sure what the amount of wattage has to do with reliability? Personally, I'd rather have a board that could handle a bit more wattage if need be than not have enough. Would you say a 400watt power supply is less reliable than a 250watt? 3) Cases. Have you been able to find a reasonably priced case for mini-itx that doesn't look like some cheap home theater appliance? I haven't. One thing often looked for (especially in the embedded space) is for the device to look like an appliance. People are much less likely to mess with something when they don't know what it is. With a mini-itx case with upfront firewire and line-out, my 14 year old cousin would have his fingers in that case in a minute! You are right here, and they are not many good cases to choose from --- YET! :-) My company has already submitted plans to a few machineshops to build some prototype ITX cases as we speak. We just sent them in last week, so it'll be a few weeks. If anyone has an suggestions on the case style or anything they would/wouldn't like to see on a mini-ITX case, please speak now before we hit full production. We will be selling them to everyone, so if there is something you've been wanting in a mini-ITX, email me ASAP so we can look at possibly adding it to our prototype. When the 7501 comes out later this year there won't even be a point of arguing this anymore. That board is going to be killer! If the 7501 can perform to the degree we need, then you could be right. :-) Your point was not missed, but I don't think it is a good idea to include that much hodge podge functionality (web server, mail server, PBX, streaming media server, etc, etc) in one system. Also, most of my customers want reliability. Which the Soekris has over the ITX stuff, hands down. It depends on your market. Our market was for the small/home office with up to about 12 users, and they would like the biggest bang for their buck. If you could sell them one piece of hardware that could do everything they need, such as DSL PPPOE client, VPN, firewall, Intrusion Detection, web/email services, voicemail streaming to windows media/real player, plus full PBX options - it makes a nice little package. Of course, they don't have to use every feature there - they could always use a WRT54G for a DSL router/firewall, and only use our appliance for what they want/need, but at least they have the option/choice. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home
Can any one tell me what the mysql password, no it's not password.. Try passw0rd with a disgit ZERO not o __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiples broadvoice lines {Scanned}
Sorry Everyone, My mother past away this week. I see there might be some fixes for this. I will try them tonight. Thanks, David On Thu, 2005-05-26 at 11:14 -0700, trixter http://www.0xdecafbad.com wrote: On Thu, 2005-05-26 at 12:48 -0500, Jay Milk wrote: Nothing wrong with putting them all in the same context and using Goto -- in fact, I've been using that with nine SIP lines from three different providers and a dozen incoming DIDs from two IAX providers. Why, you ask? Because you have your ALL call-distribution nicely contained in a single file -- extensions.conf. I never said there was anything wrong with that if that is what you choose to do, however I did say that if you do not choose to put them all in the same context and have them all go to different contexts instead asterisk ignores your feeble request and does what it wants. And that in my book qualifies as a bug. If I set a unique context for each account, the mere fact they are all from the same sip proxy should not override that. It does not if they are from different proxies so it makes no sense that it does when they are the same proxy. I think it was either a lazy programmer or a bad sort algorithm (perhaps an if that doesnt have enough compares for unique connection information?) Granted this is a rare occurance for testing purposes, if a test case was not created to test for this problem specifically it would not be uncovered until someone used asterisk to try to do exactly this. I just feel that people should have choice, simple little freedoms to do their extensions.conf however they want, and not be forced to put them all in the same context if they do not want to. Maybe my feelings on freedom and choice are too far out there and the better solution is to do it one way because that way is best for one person. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Soekris
Kristian Kielhofner wrote: Dustin Wildes wrote: Maybe my point was missed. Hardware wise - a VIA MII EDEN based board will greatly outperform a Soekris system, which is why my embedded platform is based on the VIA hardware instead of the Soekris, because I AND my customers did want an all-in-one system, and small offices tend to want an all-in-one piece of equipment. Dustin, Yes, a VIA Eden board will greatly outperform a Net4801. My CL1 is actually quite powerful. However, several points on the mini-itx architecture that need to be mentioned: 1) Heat/Reliability. Much more heat generated, my mini-itx system has three fans. The Soekris has none (not even a heatsink). This makes the Soekris much more reliable - no moving parts. I am using the MII 6000 (no fans) with a heatpipe to replace the embedded heatsink - pushing to extruded fins. It does get warm, but not that bad. 2) Power usage. All though I have yet to measure it, my mini-itx system has a 90 watt power supply (which is ATX based, btw). My Soekris has a 12 watt power supply. Also on another note of reliability, I trust the Soekris power supply much more than the half breed ATX in most mini-itx systems. Yes, I do know that just because you have a 90 watt power supply you are not using all 90 watts, but the fact that the Soekris has a 12 watt power supply means that it is DEFINITELY not using more than 12 watts. I haven't measured the power either - but we have been using the morex power supplies for several months now, and no problems. But I not sure what the amount of wattage has to do with reliability? Personally, I'd rather have a board that could handle a bit more wattage if need be than not have enough. Would you say a 400watt power supply is less reliable than a 250watt? I'm not saying that at all. But in my years of dealing with PC's, the most common things to go are the HD and the power supply. Especially ATX power supplies... I feel that the mini-itx using ATX power supplies reduces the overall reliability of the system. 3) Cases. Have you been able to find a reasonably priced case for mini-itx that doesn't look like some cheap home theater appliance? I haven't. One thing often looked for (especially in the embedded space) is for the device to look like an appliance. People are much less likely to mess with something when they don't know what it is. With a mini-itx case with upfront firewire and line-out, my 14 year old cousin would have his fingers in that case in a minute! You are right here, and they are not many good cases to choose from --- YET! :-) My company has already submitted plans to a few machineshops to build some prototype ITX cases as we speak. We just sent them in last week, so it'll be a few weeks. If anyone has an suggestions on the case style or anything they would/wouldn't like to see on a mini-ITX case, please speak now before we hit full production. We will be selling them to everyone, so if there is something you've been wanting in a mini-ITX, email me ASAP so we can look at possibly adding it to our prototype. Now we're getting somewhere! I'll get to you about this off-list... When the 7501 comes out later this year there won't even be a point of arguing this anymore. That board is going to be killer! If the 7501 can perform to the degree we need, then you could be right. :-) Hopefully I am. At this point they will probably have a mobile Athlon 64. That will SMOKE a C3 (which aren't that great to begin with). Your point was not missed, but I don't think it is a good idea to include that much hodge podge functionality (web server, mail server, PBX, streaming media server, etc, etc) in one system. Also, most of my customers want reliability. Which the Soekris has over the ITX stuff, hands down. It depends on your market. Our market was for the small/home office with up to about 12 users, and they would like the biggest bang for their buck. If you could sell them one piece of hardware that could do everything they need, such as DSL PPPOE client, VPN, firewall, Intrusion Detection, web/email services, voicemail streaming to windows media/real player, plus full PBX options - it makes a nice little package. Of course, they don't have to use every feature there - they could always use a WRT54G for a DSL router/firewall, and only use our appliance for what they want/need, but at least they have the option/choice. As of now, AstLinux has everything but IDS, VM streaming, and the e-mail server. It has a web server, but I would never want my internet facing web server to be running on the same machine as my PBX! Ditto for e-mail. As far as VM streaming, I don't really see what the point is (unless you have EXTREMELY long voicemails). If you are using wav49 you could probably download and playback the entire VM just as quickly. Other than that, it sounds
R: R: R: [Asterisk-Users] AT-320 + supervised transfer
I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: lunedì 30 maggio 2005 18.40 A: asterisk-users@lists.digium.com Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer On Monday 30 May 2005 17:22, Giordano Grandis wrote: The procedure that will do asterisk is very nice ;) but whe it was available ? Asterisk's atxfer support is only in CVS. Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. You *must* be using a new firmware for the phone. Download 1.43 from http://www.aredfox.com/edownloadssip.htm (the AT-320 needs PA186S code) Here my sip.conf for the phone, can u say me if there is somethingh wrong ? Looks fine to me.. I think is ok, maybe i have some problem on phone settings.Can I see your exmple phone setting ? They're at work so I can't see the config right now... but they're just the defaults with the DTMF changed to RFC2833 and the NTP server set... Try resetting to defaults using the procedure at http://www.voip-info.org/wiki-ATCOM+AT-320 Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] where can i get a vanity DID?
From what I understand if I have an asterisk pbx set up, I can also get a vanity DID. 1) Where I get the DID from does not matter, my voip provider can use the DID i ask them to. Is this correct? 2) What place has good vanity DID's? 3) Do toll free DID's save me or the person calling me money? Or are toll free DIDS more expensive? 4) Where can I get a good vanity DID. I would like some kind of search interface that will suggest alternate names. Thanks, Tom __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home
Can any one tell me what the mysql password, no it's not password.. Try passw0rd with a disgit ZERO not o Try amp109 __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] News From Astricon
link doesnt work - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 22, 2004 8:28 PM Subject: [Asterisk-Users] News From Astricon We've got some replies to questions online about Astricon and we now have a mirror available at: http://astricon.voctel.com/news.php If anyone has any comments about Astricon, please forward them to me and I will put them up on the site so that all the people who didn't go can read them. Cheers, Matt Riddell http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT100 Phone Died During Call
You are lucky it is still working at all. I have seen a very high number of the phones die alltogether. - Original Message - From: Jim Duda To: asterisk-users@lists.digium.com Sent: Sunday, May 29, 2005 7:44 AM Subject: [Asterisk-Users] BT100 Phone Died During Call I've been using Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000 for all my analog phones. All has worked rather flawlessly, until today. I was on the BT100 phone today. During my phone conversation, the BT100 disconnected and went into a "click" mode. 2 "clicks" per second I think. Asterisk was fine, I picked up one of the analog phones, recieved dial-tone, and was able to call out through my service. The MENU key on the BT100 would work as I was attempting to "reboot" the phone. I had to give the phone a hard power-cycle to restore it to normal. Has anyone experienced this problem with a BT100? Jim ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Urgent Help neededt!! Asterisk 1.0.7 CPU at 99%
Hi, I am using Asterisk 1.0.7 that comes with [EMAIL PROTECTED] 1.0 ISO. I have installed chan_unicall.c and MFCR2 support with latest Steve Underwood code unicall-0.0.2pre2 (yes this is the latest version, not 0.0.2pre19!!) Asterisk process is keeping the cpu at 99% most of the time. I have two Digium cards and all of them have their unique IRQ, so there is no irq conflict. [EMAIL PROTECTED] asterisk]# cat /proc/interrupts CPU0 0: 381697 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 1 XT-PIC serial 7:3773994 XT-PIC wctdm 8: 1 XT-PIC rtc 11: 230543 XT-PIC eth0 12:3780190 XT-PIC wcte11xp 14: 19066 XT-PIC ide0 NMI: 0 ERR: 0 When I make 1 call, some times the CPU spikes to 90% but I did not realize this until there are 2, 3 or more calls at the same time, the cpu keeps at 99% with audio being scrambled and unintelligible. Any help is greatly appreciated as this system entered production today and the customer is getting upset. Kind regards. Andres. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk@home
thx -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: 30 May 2005 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Can any one tell me what the mysql password, no it's not password.. Try passw0rd with a disgit ZERO not o Try amp109 __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pridialplan prilocaldialplan
Hi list! What exactly is the meaning / function of the pridialplan prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pridialplan prilocaldialplan
It defines the number pattern that is sent to the PRI. 90% of the time, best practice is to set this to 'unknown' and then Asterisk will dump the dialled digits to the PRI as the user dials them, not in some predefined pattern, UNLESS your telco expects digits to be dialled in a certain pattern. hth -Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED] Sent: Monday, May 30, 2005 12:41 PM To: Asterisk Users List Subject: [Asterisk-Users] pridialplan prilocaldialplan Hi list! What exactly is the meaning / function of the pridialplan prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk install error ...
Title: Asterisk install error ... Hi; It is my first time to use asterisk I have TDM400 wildcard and 4 FXO Modules when I install asterisk an error occurred Chen_zap.c 2772 : error : zt_event_dtmfdigit undeclared Can any body help why this error .. Thanks; Ghassan M. Lama' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Urgent Help neededt!! Asterisk 1.0.7 CPU at 99%
Hi, Are you sure the process consuming your CPU is Asterisk? Did you tried with different codecs? Andres Maduro wrote: Hi, I am using Asterisk 1.0.7 that comes with [EMAIL PROTECTED] 1.0 ISO. I have installed chan_unicall.c and MFCR2 support with latest Steve Underwood code unicall-0.0.2pre2 (yes this is the latest version, not 0.0.2pre19!!) Asterisk process is keeping the cpu at 99% most of the time. Sebas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Peer to Peer calls
If I have both clients and Asterisk in the same nat, it is working fine with internal addressing. When using outside IP with appropriate ports open(5060,1-2000), with following call flow, X-Lite-asterisk -- nat-- - as5400-pstn canreinvite=no nat=yes with this setting RTP's should be between ata186 and astersik over nat. I can see bi-directional RTP streams in Ethereal ( *asterisk-x-lite* ), very few of them from Asterisk to X-Lite, resulting one-way audio, and the call is disconnected abruptly after that. I have setup g711, on X-Lite and SIP.conf, but still it is negotiating gsm with AS5400. Eventually I wan to use clients on different nats, to work with Asterisk on different nat. Is this a codec issue, or asterisk problem or nat? can some body help, probably I need proxy. Obaid Siddiqui. Network Engineer, Prizm Communications, LP Austin, Texas. - Original Message - From: Michael J. Tubby G8TIC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 29, 2005 7:13 PM Subject: Re: [Asterisk-Users] Peer to Peer calls - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, May 29, 2005 10:41 PM Subject: Re: [Asterisk-Users] Peer to Peer calls On 00:32, Mon 30 May 05, Cenk Yabas wrote: Can anybody please answer this. Both clients are behind different NAT's. One of them starts a SIP call to the other through Asterisk. Asterisk sets up the call. Issues reinvite and connects them together. After this point does the media stream flow through Asterisk or Peer to Peer? Does such a call use any system resources of Asterisk server after connection? Thank you in advance. Did you test this ? My experience is the 'reinvite' does not work in the setup you descripted. I always have to set 'canreinvite=no' in asterisk config or the audio will not come through. If you have only one phone on both NAT's and you can do port-forwording on both firewalls, it can work, but that scenario is highly uncommon. The audio stream is setup on some random port, so your firewall will block this by default. *But* If your firewall is SIP-aware - for example a Cisco 837 ADSL router with IOS 12.3 - then it should be able to fix up the firewall rules dynamically so that when the phones in the inside (behind the firewall) re-invite it should inspect the SIP on udp/5060 and see the invitation and open the appropriate UDP port(s) for the RTP stream. Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transcoding prevention
Hi, my setup is like: phones (g729/g711)--(SER)-- Asterisk --(oh323)--gateway (supports g729g711) problem begin when phone supports only g711 and Asterisk doesn't negotiate this codec in full path (from phone to gateway), but tries to do transcoding (and because I haven't g729 codec in asterisk, the call fail). Is there any solution how to tell to Asterisk to negotiate codec, that is supported by both parties? I read some posts like my problem, is any new recomendation or devel. progress so solve? thanks PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridialplan prilocaldialplan
On Mon, 30 May 2005, Remco Barende wrote: What exactly is the meaning / function of the pridialplan prilocaldialplan? Both set the two fields Type Of Number (TON) and Numbering Plan (NPI) markers on an outgoing isdn call. These two tell a receiving isdn switch how to interpret the accompanying digits. The pridialplan option controls the TON+NPI associated with the Called Party information element. This is the recipient of the call. The prilocaldialplan option controls the TON+NPI associated with the Calling Party information element. This is the originator to be presented to the receiving user (think CallerId). I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? unknown : set TON to unknown and NPI to unknown. This instructs the receiving switch to interpret the digits according to the standard used by the pstn in that country, leading zeroes etc included. E.g. 00461234567 for a call to Sweden. This is what one should normally use local: Almost never used unless requested by your pstn provider. national : Interpret the digits as a national number, i.e. with an area code at the beginning, but without any escape digits. I.e. no leading zero or similar for the area code. International dial is not possible. This is the default in Asterisk and almost always wrong. In some pstn networks in the USA this is actually interpreted like unknown above and not according to the specification. international : A fully formed E.164 phone number. 461234567 would be a call to Sweden. Usable. dynamic : Parse the dialed number and try to find a matching prefix in the settings internationalprefix, nationalprefix, localprefix, privateprefix, unknownprefix. If matched, set the TON/NPI to the matched setting and strip the prefix. Having set nationalprefix=0 allows you to call Dial with e.g. 0461234567 and have it sent as TON=national, digits=461234567. Setting the *prefix variables listed in dynamic above will _add_ the prefix on inbound calls. This can make the parsing of incoming calls easier. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Urgent Help neededt!! Asterisk 1.0.7 CPU at 99%
This is usually a mpeg123 problem. try removing the moh module and see if it goes away. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sebastian Silva Sent: Monday, May 30, 2005 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Urgent Help neededt!! Asterisk 1.0.7 CPU at 99% Hi, Are you sure the process consuming your CPU is Asterisk? Did you tried with different codecs? Andres Maduro wrote: Hi, I am using Asterisk 1.0.7 that comes with [EMAIL PROTECTED] 1.0 ISO. I have installed chan_unicall.c and MFCR2 support with latest Steve Underwood code unicall-0.0.2pre2 (yes this is the latest version, not 0.0.2pre19!!) Asterisk process is keeping the cpu at 99% most of the time. Sebas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Ext on IP500
Sorry if this show up twice, yesterdays post did not show up. We are having trouble setting up two ext's on one phone. We have it the point where the first two lines are ext 4000 then the third line is ext 4013. We can receive calls to both ext's but we can't make out going calls on ext 4013. The other thing that is strange is we need to have ext 4013 at the top of the config for ext 4000 to make outbound calls. If we swap then 4013 can make calls but 4000 can't. Has anyone done this successfully? Rick Here is our current sip config. [4000] username=4000 type=peer secret=1234 qualify=no port=5060 pickupgroup= nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid=Rick 4000 allow=all [4013] username=4013 type=peer secret=1234 qualify=no port=5060 pickupgroup= nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid=support 4013 allow=all ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Ext on IP500
I think the first thing you have to do is to set each extension to a different port.. Line 1 Port 5060 ... Line 2 ... 5061, etc. The way this has worked for me is this: Line 1 is configured with 4000... Line 2 would not be configured in the phone.. Line 2 defaults to a dup of line 1. You would then configure Line 3 with 4013. Line one would be 5060 Line 3 5061. Bill On 5/30/05, Rick Baranowski [EMAIL PROTECTED] wrote: Sorry if this show up twice, yesterdays post did not show up. We are having trouble setting up two ext's on one phone. We have it the point where the first two lines are ext 4000 then the third line is ext 4013. We can receive calls to both ext's but we can't make out going calls on ext 4013. The other thing that is strange is we need to have ext 4013 at the top of the config for ext 4000 to make outbound calls. If we swap then 4013 can make calls but 4000 can't. Has anyone done this successfully? Rick Here is our current sip config. [4000] username=4000 type=peer secret=1234 qualify=no port=5060 pickupgroup= nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid=Rick 4000 allow=all [4013] username=4013 type=peer secret=1234 qualify=no port=5060 pickupgroup= nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid=support 4013 allow=all ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk install error ...
wich version are you trying to install? and how? wich GNU/Linux distro? did you already installed zaptel? best regards On 5/30/05, Ghassan Lama [EMAIL PROTECTED] wrote: Hi; It is my first time to use asterisk I have TDM400 wildcard and 4 FXO Modules when I install asterisk an error occurred Chen_zap.c 2772 : error : zt_event_dtmfdigit undeclared Can any body help why this error .. Thanks; Ghassan M. Lama' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nntp access
On Mon, May 30, 2005 at 06:32:34PM +0200, Marcin Kuczera wrote: hi, is it possible to get to this group via nntp ? Yes. http://gmane.org/find.php?list=asterisk -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Ext on IP500
I don't believe any of that works on a Polycom 500. Each Line appearance can handle two concurrent calls, so they don't roll over properly. The best way I have found, if you need that many extensions, is to allocate one extension per button and roll them over in your dialplan. exten = 123,1,Dial(SIP/123,5) exten = 123,2,Dial(SIP/124,5) exten = 123,3,Dial(SIP/125,5) exten = 123,4,VoiceMail,u123 exten = 123,105,VoiceMail,b123 Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Meridian 808 Function
Message: 19 Date: Mon, 30 May 2005 11:12:13 -0500 From: Carlos Chavez [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Meridian 808 Function To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On Mon, 30 May 2005 08:53:12 -0700 (PDT), Miguel Ruiz Velasco Sobrino wrote Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx programation)? Thanks for the hint The way to use is (which I do not really recommend) is that the nortel user will dial the extension that is connected to the Asterisk server and then press the FUNCTION button on his/her phone followed by 808. After 1 or 2 seconds the phone will say Long Tones. You can then dial the extension you want on the * server. The best way to integrate with a Meridian is to use FXS ports connected to trunk line ports and then configure the PBX to dial something like 7 to get a dial tone from the Asterisk server. That way your users do not have to remember a complicated procedure. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 Thanks for the tip, it worked as seen on TV! I also think the users are [somewhat] uncoperative and memoryless, but i can't connect FXS to trunk because that pbx was already full. Also having some problems with disconnect supervision. I know FXO is a hassle, but had no other means of solution. Thanks Miguel Ruiz Velasco Version: OpenKeyServer v1.2 Comment: Extracted from belgium.keyserver.net Signature: 0x59831109 __ Do you Yahoo!? Yahoo! Small Business - Try our new Resources site http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error in Zapata Config?
When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 2, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 3, FXS Kewlstart signalling -- Reconfigured channel 4, FXS Kewlstart signalling Here is my config /etc/asterisk/zapata.conf [channels] context=internal signalling=fxo_ks rxgain=0 txgain=20 channel = 1 context=analog2 signalling=fxo_ks rxgain=0 txgain=0 channel = 2 context=pstn signalling=fxs_ks faxdetect=incoming echotraining=800 echocancel=128 echocancelwhenbridged=yes relaxdtmf=yes rxgain=10 txgain=0 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes useincomingcalleridonzaptransfer=yes callerid=asreceived group=1 ;channel = 4 channel = 3,4 Thanks for any help, Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
OT: Date off??? (was Re: [Asterisk-Users] News From Astricon)
On Thu, September 23, 2004 8:41, Steve Totaro said: link doesnt work You may want to take a look at your system settings, as I think it unlikely that this e-mail has been in transit for approx 8 months... ;-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in Zapata Config?
If you mentioned it, I missed it. I'm assuming you are trying to use a digium TDM-fxo card? When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 2, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 3, FXS Kewlstart signalling -- Reconfigured channel 4, FXS Kewlstart signalling Here is my config /etc/asterisk/zapata.conf [channels] context=internal signalling=fxo_ks If it is a TDM card with fxo modules, then the above should be fxs_ks, and the same for the ones below. rxgain=0 txgain=20 channel = 1 context=analog2 signalling=fxo_ks rxgain=0 txgain=0 channel = 2 context=pstn signalling=fxs_ks faxdetect=incoming echotraining=800 echocancel=128 echocancelwhenbridged=yes relaxdtmf=yes rxgain=10 txgain=0 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes useincomingcalleridonzaptransfer=yes callerid=asreceived group=1 ;channel = 4 channel = 3,4 Thanks for any help, If you're not using a TDM card, then tell us what it is that you're trying to configure. Might also include /etc/zaptel.conf in the next post. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I865, HFC-S etc.
Hi, I'am having some problems with new mainboards and 3xHFC-S cards. The the first problem was with interrupts, I mean if HFC-S card was using interrupt i.e. 21 or higher - it didn't work. Solved by disabling APIC. However, still the driver behaves a little bit strange. If card 0 1 is TE and 2 is NT, TE works fine, but NT is not working at all. If card 0 is NT and 1 2 TE - all of them works fine. When all in TE mode - works fine. brustuff is 0.2.0-RC7k Anybody knows what can be the reason ? I used florz path - helped me to make zaphfc driver running with APIC turned on. However NT mode didn't work @ all, maybe because of my asterisk was in a different version than the florz path (RC7k , RC8a). Putting bri_net_ptmp in config hanged up asterisk. Probably this is a problem of different releases, however is there anybody having asterisk package with bristuff RC8a for DEBIAN ? Or just a short howto built it ? Or maybe there is RC7k florz patch ? Building packages is not yet my strong side ;-( Regards, Marcin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
