RE: [Asterisk-Users] IAX dns lookups
Hi,
Try something like these, works for me.
extensions.conf
[general]
;
static=yes
;
writeprotect=no
;
[globals]
;
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;
#include globals.conf ;This includes your conf file with your fqdn's
listed.
exten = _20XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _20XX,2,Hangup
;
exten = _21XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _21XX,2,Hangup
;
exten = _22XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _22XX,2,Hangup
;
exten = _23XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _23XX,2,Hangup
;
exten = _24XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _24XX,2,Hangup
globals.conf
RMT1=www.domain1.zzz;remote1
RMT2=www.domain2.zzz;remote2
RMT3=www.domain3.zzz;remote3
RMT4=www.domain4.zzz;remote4
RMT5=www.domain5.zzz;remote5
I never reboot even when the DynDns changes.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Liaan vd
Merwe
Sent: 03 February 2005 07:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX dns lookups
Hi all
Do any of you know i can force asterisk to lookup ip
addresses for peers and
trunks everytime it tries to make a call.
One of the peers has a dynamic ip and is using DynDNS
to register host. Now
i need to reload asterisk everytime i want to call it
thanks
liaan
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[Asterisk-Users] Re: problem in compiling asterisk-addons
prbolem still there first of all i have these two(asterisk, asterisk-addons) working on my system i got these packages from asterisk.org then i recompiled asterisk-addons because i want res_config_mysql.so module for real time database i got this addon by following command cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot login (password is 'anoncvs') cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co asterisk-addons -- when i recompiled this code i got these following errors -- [EMAIL PROTECTED] asterisk-addons]# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given make -C format_mp3 all make[1]: Entering directory `/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c format_mp3.c: In function `load_module': format_mp3.c:335: warning: passing arg 5 of `ast_format_register' from incompatible pointer type gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/asterisk-addons/format_mp3' cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:269: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 rm app_saycountpl.o [EMAIL PROTECTED] asterisk-addons]# ls --- i have recompiled these two packages but same result any on have fixed this problem kindly answer me i have checked the code of app_addon_sql_mysql.c and fixed the function 'AST_LIST_REMOVE' call then again execute make now this error --- [EMAIL PROTECTED] asterisk-addons]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c cc -shared -Xlinker -x -o app_addon_sql_mysql.so app_addon_sql_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o res_config_mysql.o res_config_mysql.c res_config_mysql.c: In function `realtime_mysql': res_config_mysql.c:142: warning: passing arg 1 of `ast_strlen_zero' makes pointer from integer without a cast res_config_mysql.c:144: warning: assignment makes pointer from integer without a cast res_config_mysql.c:149: warning: assignment makes pointer from integer without a cast res_config_mysql.c: In function `realtime_multi_mysql': res_config_mysql.c:177: error: storage size of `ra' isn't known res_config_mysql.c:189: warning: assignment makes pointer from integer without a cast res_config_mysql.c:250: warning: assignment makes
RE: [Asterisk-Users] howto answer a call in a queue
Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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 ___ 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 dns lookups
thanks.. will give it try.
cheers
L
- Original Message -
From: David J Carter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 03, 2005 10:07 AM
Subject: RE: [Asterisk-Users] IAX dns lookups
Hi,
Try something like these, works for me.
extensions.conf
[general]
;
static=yes
;
writeprotect=no
;
[globals]
;
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;
#include globals.conf ;This includes your conf file
with your fqdn's
listed.
exten = _20XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _20XX,2,Hangup
;
exten = _21XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _21XX,2,Hangup
;
exten = _22XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _22XX,2,Hangup
;
exten = _23XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _23XX,2,Hangup
;
exten = _24XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _24XX,2,Hangup
globals.conf
RMT1=www.domain1.zzz ;remote1
RMT2=www.domain2.zzz ;remote2
RMT3=www.domain3.zzz ;remote3
RMT4=www.domain4.zzz ;remote4
RMT5=www.domain5.zzz ;remote5
I never reboot even when the DynDns changes.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Behalf Of Liaan vd
Merwe
Sent: 03 February 2005 07:38
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] IAX dns lookups
Hi all
Do any of you know i can force asterisk to lookup ip
addresses for peers and
trunks everytime it tries to make a call.
One of the peers has a dynamic ip and is using
DynDNS
to register host. Now
i need to reload asterisk everytime i want to call
it
thanks
liaan
__
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Re: [Asterisk-Users] IAX dns lookups
On Wed, 2005-02-02 at 23:37 -0800, Liaan vd Merwe wrote: Hi all Do any of you know i can force asterisk to lookup ip addresses for peers and trunks everytime it tries to make a call. One of the peers has a dynamic ip and is using DynDNS to register host. Now i need to reload asterisk everytime i want to call it I have a system with 3 servers in various locations all on dynamic IPs, when their IP changes /etc/ppp/ip-up.local has the command to reload asterisk so that they re-register with all their peers. Works for me (tm). -- Dave Cotton [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
[Asterisk-Users] Special error numbers
Are there any special numbers in the UK that are not available busy congested etc so that I can test to make sure that I am playing the correct tones back to the user ? At the moment, I am getting a boatload of reorder / not available when I know that the line was engaged or busy. Trouble is trying to track it down because the line is not always engaged. Does BT have any special numbers to call ? Julian ___ 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] HEEEELP!!!!!!!! with file CODEC_G729.SO
On 3 Feb 2005, at 01:56, Ing. Ignacio Ortega A. wrote: Hello everyone can anyonone of you send me the file codec_g729.so this file has to be inserted in /urs/lib/asterisk/modules You can download it from Digium You then need to buy (or conceiveably beg) a license key from them. You then have to run the license registration program on your asterisk system, AND that system has to be able to make an outgoing TCP connection to Digium. Due to a patent on g729 you won't find a _legal_ free implementation. Tim. Thank You ___ 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 http://www.westhawk.co.uk/ ___ 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] load balancing 20 asterisk servers
I beleive what you're looking for is a scalable SIP proxy, like SER :) That way, all clients registers to SER and SER redirects the caller to one of the asterisk boxes. Search the wiki at voip-info.org for asterisk at large :) Yes, that is one of the many pages I've read. But we still have a problem. Take a look at this image to get a better idea of my end goal. http://drmac.homeunix.net/images/load_balancer.jpg You won't need the second balancer. SER can do that. For growth, all you do is add more SER and more Asterisk boxes. Are you sure one SER box won't be sufficient? But if Asterisk won't work correctly with the load balancing due to packet movement, then I need to approach this differently. perhaps setting up a second SER box for failover will do? just failover with heartbeat or something... roy ___ 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: volume too low.
Ousmane == Ousmane Doukara [EMAIL PROTECTED] writes: Ousmane Hi, I am trying to figure out why my recorded files have a Ousmane very low volume ? I tried gsm, wav with no success. Where do the recordings come from? If this is from a Zap channel, try increasing rxgain in zapata.conf. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ 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 7940 [SIP], DTMF and Voicemail
Hi Doug, Thanks very much - sorry I didn't see this in the wiki. The details you quoted worked perfectly and I'm now on firmware V7. I still have the no DTML tones recognised in voicemail but thats another problem. All the best! Derek Doug Lytle wrote: Derek Conniffe wrote: Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). Derek, I found this in the comments field on the wiki, hope this helps on your upgrade, it did for me, I got the same error upgrading from 6.2 to 7.3. It asked for the same files and then got the error invalid protocol whatever. Someone else with these phones gave me the solution. When you upgrade the firmware you need to have the image_version number in 2 files. The os79xx file and the sipdefault file. In all previous upgrades the version was exactly the same example P0S30203 in the OS79XX.txt and image_version P0S30203 in the sipdefault.cnf file. However when upgrading to 7.3 you need 2 different names. in OS79XX.txt you need to write P003-07-3-00 and P0S3-07-3-00 in SIPDefault.cnf then you just need the 4 files in the TFTP server directory, P003-07-3-00.bin, P003-07-3-00.sbn, P0S3-07-3-00.loads, P0S3-07-3-00.sb2. Also if you are upgrading for the 1st time to the inital version P0S30203 and it doesn't work, you make have an error in the status messages section of the phone about a buffer overflow. This can be solved by deleting everything out of sipdefault.cnf except for the image_version: P0S30203 line. Once you have loaded the firmware you can add all the other stuff. This is some buffer overflow error that doesn't let the phone download a file that is over a certain kb in size. Damian ___ 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 -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.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
[Asterisk-Users] Newbie - Echo on Incoming calls
Hi, I have a setup of Asterisk with TDM13B card (1FXS,3FXO ports) and acouple of Cisco 7960 IP phones on SIP. So far i have managed to get the internal SIP calls going through without a hitch and can get to call through to the analog lines (SIP -- ANALOG) without any echos. However whenever we receive an incoming call via the analog lines (ANALOG -- SIP), there is an echo for about 15-30 seconds and it fades off thereafter Initially i had an echo on both calling and receiving until i enabled the echotraining=yes option and that seemed to have cleared the outgoing calls. I cant seem to establish where the problem is. Please see my zapata.conf file below. PS: I have tried various values i.e FXS_KS and FXS_LS, also the tx and rx values as well but none seem to be working. Any help will be highly appreciated. Regards, ; ; Zapata telephony interface ; ; Configuration file [channels] context=default switchtype=national signalling=fxo_ks callerid=asreceived usecallerid=yes cidsignalling=v23 hidecallerid=no useincomingcalleridonzaptransfer=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1-4 immediate=no busydetect=yes busycount=7 musiconhold=default channel = 1 signalling=fxs_ls group=2 context=incoming channel= 2-4 On Thu, 2005-02-03 at 12:28, tim panton wrote: On 3 Feb 2005, at 01:56, Ing. Ignacio Ortega A. wrote: Hello everyone can anyonone of you send me the file codec_g729.so this file has to be inserted in /urs/lib/asterisk/modules You can download it from Digium You then need to buy (or conceiveably beg) a license key from them. You then have to run the license registration program on your asterisk system, AND that system has to be able to make an outgoing TCP connection to Digium. Due to a patent on g729 you won't find a _legal_ free implementation. Tim. Thank You ___ 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 http://www.westhawk.co.uk/ ___ 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 -- Michuki Mwangi KENIC. ___ 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: using the MYSQL command to insert a record
In article [EMAIL PROTECTED],
Robert Howard [EMAIL PROTECTED] wrote:
I am trying to use the MYSQL command to insert a
record into a database and I can't seem to get it to
work. I can do an UPDATE with no problem.
Here is the line in my dialplan
exten = s,12,MYSQL(QUERY resultid ${connid} INSERT
INTO `member` ( `id` , `member_num` , `active` )VALUES
('',${number}' , '1'))
Does anyone have an example of an INSERT INTO that I
could look at?
Firstly, backticks around table and column names are not needed unless
the name has weird characters in it.
Secondly, the whole query needs single quotes around it.
Thirdly, single quotes within the query need \-escaping.
Here's a working example which updates a database on hangup:
exten = h,1,MYSQL(Connect conn localhost username password dbname)
exten = h,2,MYSQL(Query res ${conn} 'INSERT INTO
responses(adv_id,resp_callerid,resp_file,resp_created)
VALUES(${advertid},\'${CALLERIDNUM}\',\'${RECORDED_FILE}\',NOW())')
exten = h,3,MYSQL(Disconnect ${conn})
Hope this helps!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] IAX2 Rejected connection attempt (voiptalk.org)
Hi all, Have a problem that I have been battling with for a few days now with help from voiptalk.org support.but I thought someone here might have seen this before. I have an asterisk box running on a real non nat'ed ip address with an incoming number from voiptalk.org on IAX2. The problem I am seeing with or without firewall rules in place (port 4569 udp open or all ports open ie firewall rules flushed) is rejected connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server... The real killer here is that this was all working but my inter asterisk sip connections were not transmitting voice, after changing canreinvite=no to canreinvite=yes in sip.conf the sip connections started sending/receiving voice ok but the incoming calls were being rejected. I can also dial out on voiptalks IAX connection happily. And just to be sure I changed the canreinvite back the problem remains. I have all but rebooted the box. I have restarted asterisk and checked everything again, but I just can't see why this is happening. Cheers, Mark ___ 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: load balancing 20 asterisk servers
The DNS approach does not handle single or multiple system failures, only very elementary load balancing over a lengthy period of time. Are you shure of that? I'm aware that the load criteria is trickier, but very possible. Yes, very sure. Look at past posts relative to the Broadvoice.com problem and you'll see one step in the recommended 'fix' was to install a /etc/hosts entry in the customer's system. Once something like that is done getting that admin to remove/change it is almost impossible. Operating systems and probably a lot of devices *cache* the results of DNS lookups. That means removing A records won't do any good. One can specify a short dns cache time within the primary dns, however a substantial number of machines ignore the value. Short story: No matter what network service is being balanced, if you want to guard against failure and against customers noticing that failure use a real load balancing solution, DDNS is not suitable. Agree with that 100%. ___ 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: load balancing 20 asterisk servers
How about a management server that polls the asterisk servers every minute with snmp to check cpu and ram cache, maybe even drive space. Then you could have a script decide whether the server can handle anymore connections. There are lots of different ways to measure how busy a server happens to be and snmp can be made to do that. Another is simply counting pkts/sec to/from a server. I am still a beginner so I am not sure how you could have asterisk delegate calls to other servers. would a redirect transfer remove the management server from the loop? Part of the problem is that something needs to detect a failed server and that failure can be anything from a broken cat5 cable to an internal s/w error (* failed, OS is up), etc. using loadbalancer? http://www.vovida.org/applications/downloads/loadbalancer/ The person that wrote that has a rather lengthy list of ToDo's and hasn't touched it since May 2002. I'd bet they decide the problem was much larger then what they initially thought. Documenting and handling the exception conditions with sip rtp is more difficult then what it appears on the surface. these are just ideas I am tossing out here for you. Writing code to handle asterisk-to-asterisk load balancing would be substantially easier then dealing with sip phones and adapters that have hard coded logic, and in at least some cases, that logic is less then robust. All sip devices would need to be covered one way or another. ___ 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] Grandstream ATA 486 works only with ulaw and alaw codecs.
Does anybody has got the some problem? The grandstream ATA 486 schould support almost all codecs, but it doesn't work. I get the following message when I force the use of different codec WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs! Feb 3 11:17:15 NOTICE[9529]: chan_sip.c:7395 handle_request: Unable to create/find channel What could I do to see some more detailed logs? My sip.conf [p1] type=friend username=p1 fromuser=p1 dtmfmode=rfc2833;info;inband;info;rfc2833 ;inband info http://www.voip-info.org/wiki-Asterisk secret= host=dynamic amaflags=default; Choices are default, omit, billing, documentation allow=all I was trying to change almost anything. In the granstream configuration webpage are the following things to configure, I don't understend, maybe it could do that tric. G723 rate: 6.3kbps encoding rate5.3kbps encoding rate // tried both iLBC frame size: 20ms30ms // 20 ms iLBC payload type: (between 96 and 127, default is 98) //98 Silence Suppression: NoYes Voice Frames per TX: 2 (up to 10/20/32/64 for G711/G726/G723/other codecs respectively) // did not try Layer 3 QoS: (Diff-Serv or Precedence value) // 48 Layer 2 QoS: 802.1Q/VLAN Tag 802.1p priority value (0-7) /0 0 I'm using the 1.0.5.16 firmaware, and was using the buggy 1.0.5.21 My asterisk is working fine with about 8 SIP and IAX2 providers using any codecs ... (also the 723) I'm using about 1 month old Asterisk from the CVS. Any comments would be appreciated. ___ 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] 403 Forbidden when registering sip user database on backend
i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. -- *CLI odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI -- and user is added to sip_buddies table. -- mysql update sip_buddies set auth='plaintext'; Query OK, 1 row affected (0.00 sec) Rows matched: 1 Changed: 1 Warnings: 0 mysql select * from sip_buddies; +--+--+-+--+---+--+-+-+---+--+--++-+---+---+--+--+-+---+--++--+-+--+-+-+++++--+---+--+++---+ | uniqueid | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | host| incominglimit | outgoinglimit | insecure | language | mailbox | md5secret | nat | permit | deny | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | allow | disallow | regseconds | ipaddr | auth| +--+--+-+--+---+--+-+-+---+--+--++-+---+---+--+--+-+---+--++--+-+--+-+-+++++--+---+--+++---+ |1 | 3000 | NULL| NULL | NULL | kamran | n | test| NULL | info | NULL | NULL | dynamic | NULL| NULL | y| en | | | no | NULL | NULL | NULL| 5060 | | NULL | 60 | NULL || friend | 3000 | alaw | | 1105743045 || plaintext | +--+--+-+--+---+--+-+-+---+--+--++-+---+---+--+--+-+---+--++--+-+--+-+-+++++--+---+--+++---+ 1 row in set (0.00 sec) -- i have added values to extconfig.conf -- [settings] queues.conf = odbc,asteriskcdrdb,ast_config sip.conf = odbc,asteriskcdrdb,sip_buddies -- what is the problem with my asterisk i think table is not binded with sip.conf. because when i add user to sip.conf it is registring the user and when i remove it from sip.conf it is giving 403 frobidden can any one solve this issue thanks __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.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
[Asterisk-Users] How to forward a call to the same ISDN box ?
Hello Guys Im trying to forward a incoming call from asterisk to a second number (the second phone number is located on the same ISDN BOX ) I did try the following on the extensions.conf exten=2,1,Dial(capi/720XXX1:720XXX2,18) It does work if the second number is a phone number located outside of the building (lets say a mobile phone number).. But I don't know why I cant forward the call if the phone number is on the same ISDN box. Example: Incoming call -- 720XXX1 (Asterisk server pick's up the call) --- (forwarding to) 720XXX2.. Did anybody ever got to manage this problem ? Thank you regards from Switzeeland :) Matt ___ 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] Incoming SIP calls with different signaling and RTP IP addresses
Hello, I use Asterisk CVS-v1-0-12/21/04-11:05:29 and I noticed that when we receive calls from a partner's IP address (who has a static host entry in the sip.conf file) but the RTP comes from a different address than the signaling, our * sends a 403 forbidden message and drops the call. This problem does not llow us to receive calls from SIP proxies. Was this fixed in newer versions of Asterisk? Best regards, Vlasis. ___ 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: load balancing 20 asterisk servers
DNS based load ballancing has it's place, as dose using an application level switch. Say an earthquake takes out your California data center. Shortly the DNS servers will notice and pull that center's record. However do to caches and all this is not fast and users will notice. What the switch does is route at the protocol level between local machines. You can take a machine off line and no one will notice. Works great until the big quake a backhoe takes out a fiber cable ro there is a fire flood or who knows what. You have fiber-seeking-backhoes in your area? Wow! protocol level switches have to know about the protocol. You can buy them that work with HTTP, HTTPS and the common ones but I wonder aboit SIP? Getting the RPT to the right * server would be hard beetrer to have a proxy tell the user which * server to go to and nothave to route RTP. Handling sip-rtp via a load balancer is roughly equivalent to handling ftp (ports 20 21, passive, etc). The load balancer really needs to inspect sip packet content and follow the rtp port negotiation process. I'm not aware of any balancers that can do that today. ___ 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] How to forward a call to the same ISDN box ?
Hi! Im trying to forward a incoming call from asterisk to a second number (the second phone number is located on the same ISDN BOX ) I did try the following on the extensions.conf exten=2,1,Dial(capi/720XXX1:720XXX2,18) It does work if the second number is a phone number located outside of the building (lets say a mobile phone number).. But I don't know why I cant forward the call if the phone number is on the same ISDN box. Think a little, and think again... and you'll find that you'd need three (3) lines for this to work, and you probably only have two (since you didnt provide any info I assume you have ISDN BRI). Unless you are doing ECT stuff that is, of course, then it might work. You have/want: 1 incoming call +1 redirection = outgoing call +1 incoming call 2 as result of your redirction = 3 B-channels Cheers, Philipp ___ 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] load balancing 20 asterisk servers
Hi! http://drmac.homeunix.net/images/load_balancer.jpg You won't need the second balancer. SER can do that. Seconded. For growth, all you do is add more SER and more Asterisk boxes. Are you sure one SER box won't be sufficient? Makes sense to me to have these TWO - you can take one of those off-line without interrupting service, and that's the entire idea of this discussion, isn't it? ;- Cheers, Philipp ___ 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: load balancing 20 asterisk servers
I'm trying to stay away from a software based load balancer cause what happens if that server fails? Its far less likely for a piece of dedicated hardware to fail than an actual computer. A piece of dedicated hardware runs an OS as well. I've been running software solutions for virtually everything there is, with linux, and it's rock stable. roy ___ 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] Newbie - Echo on Incoming calls
I have a setup of Asterisk with TDM13B card (1FXS,3FXO ports) and acouple of Cisco 7960 IP phones on SIP. So far i have managed to get the internal SIP calls going through without a hitch and can get to call through to the analog lines (SIP -- ANALOG) without any echos. However whenever we receive an incoming call via the analog lines (ANALOG -- SIP), there is an echo for about 15-30 seconds and it fades off thereafter Initially i had an echo on both calling and receiving until i enabled the echotraining=yes option and that seemed to have cleared the outgoing calls. I cant seem to establish where the problem is. Please see my zapata.conf file below. PS: I have tried various values i.e FXS_KS and FXS_LS, also the tx and rx values as well but none seem to be working. Try echotraining=800 and see what happens. Don't forget to stop/start asterisk; reload won't cut it. ___ 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] load balancing 20 asterisk servers
For growth, all you do is add more SER and more Asterisk boxes. Are you sure one SER box won't be sufficient? Makes sense to me to have these TWO - you can take one of those off-line without interrupting service, and that's the entire idea of this discussion, isn't it? ;- Yeah Get two cisco load balancers. One of them _will_ fail. Put them in front of two SER boxes, crossover connected. Get a gigabit switch with a good backplane Put your asterisk servers behind the SER Play :) ___ 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] Installing ASTERIS@HOME, How to install on text mode same help? {Scanned}
Hello~!
I see again this message on text mode
installing [EMAIL PROTECTED]:
"You are using unsupported hardware by
CentOS, press OK" if press OK reboot.
have aminor version, of [EMAIL PROTECTED]? to minor hardware?
Max Rivera
Brazil
- Original Message -
From:
David Shaw
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, February 02, 2005 12:57
PM
Subject: Re: [Asterisk-Users] Installing
[EMAIL PROTECTED],How to install on text mode
same help? {Scanned}
When it asked to install type "linux text"
without the "".
But when I installed my [EMAIL PROTECTED] I believed it just
installed..
David
- Original Message -
From:
Max
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, February 02, 2005 2:40
AM
Subject: [Asterisk-Users] Installing [EMAIL PROTECTED],How to install on text mode
same help? {Scanned}
Hello, Thanks for Help!
when try to install [EMAIL PROTECTED] powered by CEntOS
normal boot, 3 minutes latter:
"You are using unsupported hardware by
CentOS, press OK" if press OK reboot.
I increment mor ram and CPU:
CPU K6II- 500Mhz196Ram HD 20GB
Lan cart 10/100MbFax modem genius(Lucent chipset)Fax Modem
USR 33.66Sound OnBoard
Disk
Driver 1.44
CD
52X
How to install on text mode?
regards!
Max Rivera
Fprm Brazil.--
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RE: [Asterisk-Users] Reccomendation for reliable handsets
Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ 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] Re: IAX2 Rejected connection attempt (voiptalk.org)
Mark Benson [EMAIL PROTECTED] wrote: Hi all, Have a problem that I have been battling with for a few days now with help from voiptalk.org support.but I thought someone here might have seen this before. I have an asterisk box running on a real non nat'ed ip address with an incoming number from voiptalk.org on IAX2. The problem I am seeing with or without firewall rules in place (port 4569 udp open or all ports open ie firewall rules flushed) is rejected connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server... Where is this message coming from? Asterisk? kernel IPtables? Also, where is it appearing? e.g. /var/log/messages, console, or Asterisk log file. What does your iax.conf look like? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
[Asterisk-Users] Nortel i2004 support asterisk?
Hello everyone i simply just asking if the Nortel i2004 telefhone can work with asterisk if it so HOW? Thank You ___ 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] HEEEELP!!!!!!!! with file CODEC_G729.SO
Than You On Thu, 3 Feb 2005 09:28:56 +, tim panton [EMAIL PROTECTED] wrote: On 3 Feb 2005, at 01:56, Ing. Ignacio Ortega A. wrote: Hello everyone can anyonone of you send me the file codec_g729.so this file has to be inserted in /urs/lib/asterisk/modules You can download it from Digium You then need to buy (or conceiveably beg) a license key from them. You then have to run the license registration program on your asterisk system, AND that system has to be able to make an outgoing TCP connection to Digium. Due to a patent on g729 you won't find a _legal_ free implementation. Tim. Thank You ___ 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 http://www.westhawk.co.uk/ ___ 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] howto answer a call in a queue
My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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 ___ 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
Re: [Asterisk-Users] Outlook Integration
On Wed, 2005-02-02 at 20:57, Matt Riddell wrote: Dan Adams wrote: Are you going to be making this one available to all. I am not sure if Yes. or how it is possible, but maybe you would be able to have it so that if you right click on the contact, it has an option to iniate a call from Well, you drag the contact from outlook onto one of the speed dial buttons. Then when you click on the contact it checks the length of the phone number and prepends any digits necessary (i.e. a 4 digit number will be dialed as it, whereas a 7 digit number will have 9 prepended if it 9 is the predial code). there. If I may ask, trying to think how the thing you are making will interact with my asterisk server, would it use the telnet utility or some sort of .call file thing? Telnet basically to the manager interface then uses originate to create calls. Personally what I would find very usefull would be to have the application use Diax for making the voice calls. Diax's browser integration could be used as an interface. ___ 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] howto answer a call in a queue
Thnx i would let you know my results!! Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Error on compiling oh323
Hi!, I tride comile oh323. I have downloaded local versions of pwlib oh323 (both Janus patched). I get following errors on asterisk-oh323-0.6.5: for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/death/1/asterisk-oh323-0.6.5/wrapper' ./check_ver /root/pwlib pwlib ./check_ver /root/openh323 openh323 ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/home/death/1/asterisk-oh323-0.6.5/wrapper' make[1]: Entering directory `/home/death/1/asterisk-oh323-0.6.5/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/include/asterisk -I../wrapper -g -c -o chan_oh323.o chan_oh323.c In file included from /usr/include/string.h:33, from chan_oh323.c:34: /usr/lib/gcc-lib/i386-asplinux-linux/3.3.3/include/stddef.h:213: error: syntax error before typedef In file included from chan_oh323.c:34: /usr/include/string.h:38: error: syntax error before extern /usr/include/string.h:39: error: syntax error before __THROW ... -- skiped -- ___ 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: load balancing 20 asterisk servers
Rich Adamson wrote: [snip] I'm not aware of any balancers that can do that today. Afaik Cisco is working on SIP aware loadbalancer functionality. Don't know what the status is and since it's Cisco I'm sure it will cost a bundle. Regards, 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
Re: [Asterisk-Users] Newbie - Echo on Incoming calls
On Thu, 2005-02-03 at 14:24, Rich Adamson wrote: Try echotraining=800 and see what happens. Don't forget to stop/start asterisk; reload won't cut it. Thanks Rich, that seems to have cured it. ___ 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] Chan_Capi initial deadlock
Hi,
I had applied the patch and it got much better. Now I only have problems
every two days
eb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries!
Feb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries!
Feb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked:
Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries!
Any idea?
Regards
Felix
Jan 18 16:00:09 WARNING[2919]: Avoided initial deadlock for
'CAPI[contr1/1429092]/128', 10 retries!
2.) Patch to chan_capi
I did not tried it. The patch should solute that problems and enable
faxing? Has anybody experiences with it? If there is a
problem why is
not kapejod solving that?
You should try :)
If you don't want the fax support, you can just change this line :
--- original/chan_capi.c Fri Aug 13 12:07:28 2004
+++ chan_capi/chan_capi.c Wed Oct 27 18:55:32 2004
@@ -556,7 +556,7 @@
}
}
// wait for the B3 layer to go down
- while (i-state != CAPI_STATE_CONNECTED) {
+ while ((i-state != CAPI_STATE_CONNECTED) (i-state !=
CAPI_STATE_DISCONNECTED)) {
usleep(1);
}
}
kapejod is (was ?) quite unresponsive.
--
Carl
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[Asterisk-Users] Asterisk crashes from time to time
Hello List, we have 3 Asterisk boxes running under Fedora Core 2. Every box hangs/crashes from time to time. These installations are image based, means we made an image from our testserver with an image tool, which is able to manage ext3 partitions and deployed it to different server hardware. These servers run very stable and I could not find any failures in the logs. As these crashes appeared the first time, I thought rebooting these machines by cronjob every night at 04:00 would solve the problems. It seemed to work quite well for a couple of weeks. Today I saw our own asterisk production server crash :-( . These crashes are always the same, asterisk stops responding, the cli does not give any reaction on command input, you have to manually kill -9 all asterisk and moh processes. Asterisk logs are empty. We don't use any isdn/fxs/fxo/e1/t1 cards in these servers. Our connections to PSTN is only made by Patton/Inalp SmartNode Gateways, connected to asterisk via sip protocol. Scince these crashes appear on three servers with different hardware, and the main installation is always the same, I would think there are only two possible sources to find the failure: Operating System Fedora Core 2 Kernel 2.6.8-1.521 Asterisk CVS-HEAD-01/08/05 Has anybody out there similar problems, and if yes, how did he fix them? Is there any working solution, having asterisk control itself perhaps by using a script that drops a test call in /var/spool/asterisk/outgoing and if this call wasn't processed successfull the script stops all running asterisk and moh processes and restarts asterisk? Any help would be appreciated, since I can't get no sleep with these timebombs out there ;-) Guido Hecken ___ 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] Cisco 7960G phone crashes during SIP upgrade
Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) Thanks - Nicolas Chabbey [EMAIL PROTECTED] Leafnet Networking Research Laboratory http://www.bgp6.info - ___ 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 7960G phone crashes during SIP upgrade
Which sip ver are you trying to install. Is it stuck in a loop or anything? -Original Message- From: Nicolas Chabbey [mailto:[EMAIL PROTECTED] Sent: Thursday, February 03, 2005 7:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) Thanks - Nicolas Chabbey [EMAIL PROTECTED] Leafnet Networking Research Laboratory http://www.bgp6.info - ___ 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] Re: outbound 911 calling
If each company is in their own context, then just specify the callerID for each in their own company-specific outbound context with the SetCallerId command. Since you will have 2 totally different contexts, each company should be isolated to their own set of instructions and thus have 2 different callerid's set. - Pedro On Wed, 2 Feb 2005 22:31:57 -0500, Jason Brown [EMAIL PROTECTED] wrote: Pedro Exactly my point. I have each company in a different context. How do I SetCallerID to a number based on the context they are in? ___ 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] howto answer a call in a queue
I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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 ___ Asterisk-Users mailing list
Re: [Asterisk-Users] Asterisk crashes from time to time
we have 3 Asterisk boxes running under Fedora Core 2. Every box hangs/crashes from time to time. These installations are image based, means we made an image from our testserver with an image tool, which is able to manage ext3 partitions and deployed it to different server hardware. These servers run very stable and I could not find any failures in the logs. As these crashes appeared the first time, I thought rebooting these machines by cronjob every night at 04:00 would solve the problems. It seemed to work quite well for a couple of weeks. Today I saw our own asterisk production server crash :-( . These crashes are always the same, asterisk stops responding, the cli does not give any reaction on command input, you have to manually kill -9 all asterisk and moh processes. Asterisk logs are empty. We don't use any isdn/fxs/fxo/e1/t1 cards in these servers. Our connections to PSTN is only made by Patton/Inalp SmartNode Gateways, connected to asterisk via sip protocol. Scince these crashes appear on three servers with different hardware, and the main installation is always the same, I would think there are only two possible sources to find the failure: Operating System Fedora Core 2 Kernel 2.6.8-1.521 Asterisk CVS-HEAD-01/08/05 Has anybody out there similar problems, and if yes, how did he fix them? Is there any working solution, having asterisk control itself perhaps by using a script that drops a test call in /var/spool/asterisk/outgoing and if this call wasn't processed successfull the script stops all running asterisk and moh processes and restarts asterisk? Far too many variables for anyone to even guess at the root cause. Problem could be related to slight differences in o/s libraries between systems, coding problems within asterisk, etc. There were some issues reported with cvs head in January relative to hangs, etc. Might consider changing /etc/asterisk/logger.conf and add debug to the list. Then after a failure, at least look at /var/log/asterisk/debug messages. For additional info, I'd suggest compiling the code on one of thse machines to see if it complains about missing/inappropriate items. ___ 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 7960G phone crashes during SIP upgrade
I've tried different versions, like the 5.0, 6.0 SIP but it doesn't make any changes as the phone doesn't request the image, nor any others file after downloading the AS79XX.TXT. Once restarted, it do the same things, like configurating IP, requesting the load file and configurations on the TFTP and looping endless by restarting, crashing, restarting,... On Thu, 3 Feb 2005 07:38:18 -0600, Matt Schulte [EMAIL PROTECTED] wrote: Which sip ver are you trying to install. Is it stuck in a loop or anything? -Original Message- From: Nicolas Chabbey [mailto:[EMAIL PROTECTED] Sent: Thursday, February 03, 2005 7:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) Thanks - Nicolas Chabbey [EMAIL PROTECTED] Leafnet Networking Research Laboratory http://www.bgp6.info - ___ 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] Cisco 7960G phone crashes during SIP upgrade
I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) That's been covered at least a 100 times in the last year. Check the archives and wiki. Bottom line... on most models of the Cisco phones, you have to upgrade from sccp to sip in steps. Sip v2, v3, v5, etc. On some, you'll need to remove a bunch of the comments from the config files during the upgrade as you'll bump into some sort of buffer overflow that tends to suggest certain versions of firmware don't allocate enough space to read the entire config file from tftp. Check the wiki. ___ 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 7960G phone crashes during SIP upgrade
There was a bug with certain SCCP images, where you couldn't upgrade to images above XX size. I don't have the Cisco document though, that could be what you're seeing.. Their symptoms were worse though, it would fail *after* ugprading therefore making the phone useless :-) Not sure, just one phone you say? We upgraded all of ours in our office to 7.3 without a problem. -Original Message- From: Nicolas Chabbey [mailto:[EMAIL PROTECTED] Sent: Thursday, February 03, 2005 7:51 AM To: Matt Schulte Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade I've tried different versions, like the 5.0, 6.0 SIP but it doesn't make any changes as the phone doesn't request the image, nor any others file after downloading the AS79XX.TXT. Once restarted, it do the same things, like configurating IP, requesting the load file and configurations on the TFTP and looping endless by restarting, crashing, restarting,... On Thu, 3 Feb 2005 07:38:18 -0600, Matt Schulte [EMAIL PROTECTED] wrote: Which sip ver are you trying to install. Is it stuck in a loop or anything? -Original Message- From: Nicolas Chabbey [mailto:[EMAIL PROTECTED] Sent: Thursday, February 03, 2005 7:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) Thanks - Nicolas Chabbey [EMAIL PROTECTED] Leafnet Networking Research Laboratory http://www.bgp6.info - ___ 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] Re: IAX2 Rejected connection attempt (voiptalk.org)
Tony Mountifield wrote: Mark Benson [EMAIL PROTECTED] wrote: Hi all, Have a problem that I have been battling with for a few days now with help from voiptalk.org support.but I thought someone here might have seen this before. I have an asterisk box running on a real non nat'ed ip address with an incoming number from voiptalk.org on IAX2. The problem I am seeing with or without firewall rules in place (port 4569 udp open or all ports open ie firewall rules flushed) is rejected connection attempt from xxx.xxx.xxx.xxx which is voiptalks IAX server... Where is this message coming from? Asterisk? kernel IPtables? Also, where is it appearing? e.g. /var/log/messages, console, or Asterisk log file. What does your iax.conf look like? Cheers Tony The message is on the asterisk console - this is all I see even if iax2 debug is on and verbose is 30+ iax.conf looks like this (more or less - comments removed) [general] bindport=4569 allow=all ; same as bandwidth=high disallow=lpc10 jitterbuffer=no [voiptalk] type=peer username= secret=xx context=default host=iax.voiptalk.org [08700nn] type=peer username=08700nn context=default host=iax.voiptalk.org Last two items as per voiptalks' instructions (user and pass and 0870 no removed for list) ___ 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: problem in compiling asterisk-addons
How come you didn't 'cvs update' your /usr/src/asterisk/ ? What is the version of your asterisk source? -Matthew - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 2:24 AM Subject: [Asterisk-Users] Re: problem in compiling asterisk-addons prbolem still there first of all i have these two(asterisk, asterisk-addons) working on my system i got these packages from asterisk.org then i recompiled asterisk-addons because i want res_config_mysql.so module for real time database i got this addon by following command cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot login (password is 'anoncvs') cvs -d :pserver:[EMAIL PROTECTED]:/usr/cvsroot co asterisk-addons -- when i recompiled this code i got these following errors -- [EMAIL PROTECTED] asterisk-addons]# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given make -C format_mp3 all make[1]: Entering directory `/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o dct64_i386.o dct64_i386.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o decode_ntom.o decode_ntom.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o layer3.o layer3.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o tabinit.o tabinit.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o interface.o interface.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c format_mp3.c: In function `load_module': format_mp3.c:335: warning: passing arg 5 of `ast_format_register' from incompatible pointer type gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o make[1]: Leaving directory `/asterisk-addons/format_mp3' cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:269: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 rm app_saycountpl.o [EMAIL PROTECTED] asterisk-addons]# ls --- i have recompiled these two packages but same result any on have fixed this problem kindly answer me i have checked the code of app_addon_sql_mysql.c and fixed the function 'AST_LIST_REMOVE' call then again execute make now this error --- [EMAIL PROTECTED] asterisk-addons]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_saycountpl.o app_saycountpl.c cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c cc -shared -Xlinker -x -o app_addon_sql_mysql.so app_addon_sql_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o res_config_mysql.o res_config_mysql.c res_config_mysql.c: In function `realtime_mysql': res_config_mysql.c:142: warning: passing arg 1 of `ast_strlen_zero' makes pointer from
Re: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade
Matt Schulte wrote: one phone you say? We upgraded all of ours in our office to 7.3 without a problem. The 7.3 firmware has 2 files associated with it, P003-07-3-00 and P0S3-07-3-00, the OS79XX.TXT file didn't have them both listed. I put mine as such: [OS79XX.TXT] P003-07-3-00 P0S3-07-3-00 [SIPDefault.cnf] image_version: P003-07-3-00 image_version: P0S3-07-3-00 I'm not 100% sure I needed to list both images in each, maybe 1 in the OS79XX.TXT and the other in the SipDefault.cnf. I was trying everything, I turned around to see the phone upgrading during the test. I have another 7960($54 on eBay, just gotta love it) coming in on Friday and I'll find out then. Doug ___ 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 crashes from time to time
Rich, Far too many variables for anyone to even guess at the root cause. Problem could be related to slight differences in o/s libraries between systems, coding problems within asterisk, etc. You 're right, it could be every thing There were some issues reported with cvs head in January relative to hangs, etc. Are they reported in the bugtracker, or in the mailing list? Might consider changing /etc/asterisk/logger.conf and add debug to the list. Then after a failure, at least look at /var/log/asterisk/debug messages. Yes, this was the first thing, I did after the crash showed up. I simply forgot to enable it, since this production server ran long time without problems. But now, following murphy's law, the next crash will never happen ;-) For additional info, I'd suggest compiling the code on one of thse machines to see if it complains about missing/inappropriate items. After these machines were setup, we compiled new code on every machine, since we started with an older version of Asterisk in November 2004. The compiling of asterisk did not show me any relevant (?) errors. But I remember there were some statements (Warnings) in the console output of the make process, I didn't understand. Is this output logged in addition to the console in a logfile somewhere? If so, one could examine this output and hopefully get some hints... Thanks for your help Guido Hecken ___ 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] 403 Forbidden when registering sip user databaseon backend
Your problem is that you are binding 'sip.conf' to a non-conf table. Replace sip.conf with sipfriends -Matthew - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 4:40 AM Subject: [Asterisk-Users] 403 Forbidden when registering sip user databaseon backend i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. -- *CLI odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI -- and user is added to sip_buddies table. -- mysql update sip_buddies set auth='plaintext'; Query OK, 1 row affected (0.00 sec) Rows matched: 1 Changed: 1 Warnings: 0 mysql select * from sip_buddies; +--+--+-+--+---+--+- +-+---+--+--++-+ ---+---+--+--+-+---+ --++--+-+--+-+-+ ++++--+---+--+-- --++---+ | uniqueid | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | host| incominglimit | outgoinglimit | insecure | language | mailbox | md5secret | nat | permit | deny | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | allow | disallow | regseconds | ipaddr | auth| +--+--+-+--+---+--+- +-+---+--+--++-+ ---+---+--+--+-+---+ --++--+-+--+-+-+ ++++--+---+--+-- --++---+ |1 | 3000 | NULL| NULL | NULL | kamran | n | test| NULL | info | NULL | NULL | dynamic | NULL| NULL | y| en | | | no | NULL | NULL | NULL| 5060 | | NULL | 60 | NULL || friend | 3000 | alaw | | 1105743045 || plaintext | +--+--+-+--+---+--+- +-+---+--+--++-+ ---+---+--+--+-+---+ --++--+-+--+-+-+ ++++--+---+--+-- --++---+ 1 row in set (0.00 sec) -- i have added values to extconfig.conf -- [settings] queues.conf = odbc,asteriskcdrdb,ast_config sip.conf = odbc,asteriskcdrdb,sip_buddies -- what is the problem with my asterisk i think table is not binded with sip.conf. because when i add user to sip.conf it is registring the user and when i remove it from sip.conf it is giving 403 frobidden can any one solve this issue thanks __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.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 ___ 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 7960G phone crashes during SIP upgrade
Nicholas: You need to convert from SCCP to SIP by loading image_version: P0S30202 first. Use the OS79XX.TXT file to specify this version. After that upgrade to each newer release P0S30203, P0S3-03-2-00, etc in the same fashion. Going from 6.3 to 7.0 the loader process changes. You need the OS79XX.TXT file, the SIPmac.cnf and SEPmac.cnf.xml for the phone. From 7.1 on you don't need the OS79XX.TXT file anymore. Nicolas Chabbey wrote: Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) Thanks - Nicolas Chabbey [EMAIL PROTECTED] Leafnet Networking Research Laboratory http://www.bgp6.info - ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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] Reccomendation for reliable handsets
I have been using an IN1002 generic handset (supposed to be an unbranded cisco copy but I am skeptical) for a few months (6months+) now, and it seems pretty stable - however I haven't found a reliable supplier Also there is almost no support for them.. I have switched to the grandstream budgetone 102 and they seem pretty good too. You can pretty much plug in and forget it with both phones. They do lock up occasionally (once a month to once every 3 months). I have yet to upgrade the firmware on the grandstreams... Mark Brett, Gary wrote: Sorry to move this up the list again, but does anybody have any advice on this -Original Message- From: Brett, Gary [mailto:[EMAIL PROTECTED] Sent: 02 February 2005 10:49 To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Reccomendation for reliable handsets Hi there I'm sure this question has been raised a number of times before, but unfortunately I do not have direct access to the archives I am about to roll out Asterisk to a few companies and would like to hear your experiences about the various handsets/phones that are Asterisk compatible I am primarily looking for 2 options, the first being a cheaper model which will provide reliability whilst still maintaining a reasonable feature set, and a reliable model from the more expensive range with more features But the definite focus here is on reliability and ease of maintenance Any help or advice would be greatly appreciated; I would really like to hear your experiences/recommendations Cheers Gary ___ 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] Re: load balancing 20 asterisk servers
-Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] You have fiber-seeking-backhoes in your area? Wow! They're everywhere, man! When I was in college an entire nearby town lost all phone service for 24 hours due to a backhoe cutting a fiber optic cable. 3,000 people with no way of calling emergency services for an entire day. ___ 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] good god! stop the damn auto-replys!
Every freekin' time I post something to this list I get bombarded with out of office auto-replys. Is there no way to stop this? (other than not posting to the list..) -Matthew ___ 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 7960G phone crashes during SIP upgrade
Thanks for your reply. I've already seen the wiki page concerning the xml configuration file problem and mine have absolutly no comments on it, just the minimal lines needed. I've just tried to put the 'P0S30202' on OS79XX and as expected it crash. I think i'll not be able to upgrade to SIP by this way as all name begining with P0S3 or P0M (for MGCP) will automatically hang the device. On Thu, 03 Feb 2005 09:26:56 -0500, Steve Blair [EMAIL PROTECTED] wrote: Nicholas: You need to convert from SCCP to SIP by loading image_version: P0S30202 first. Use the OS79XX.TXT file to specify this version. After that upgrade to each newer release P0S30203, P0S3-03-2-00, etc in the same fashion. Going from 6.3 to 7.0 the loader process changes. You need the OS79XX.TXT file, the SIPmac.cnf and SEPmac.cnf.xml for the phone. From 7.1 on you don't need the OS79XX.TXT file anymore. Nicolas Chabbey wrote: Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) Thanks - Nicolas Chabbey [EMAIL PROTECTED] Leafnet Networking Research Laboratory http://www.bgp6.info - ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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] good god! stop the damn auto-replys!
Yep, send the out of office replies various junk files :) Everyone yesterday got some nice 4 mb photos of the snow in New York that I took last week. It's just a gentle reminder that they should learn how to use technology before they implement it against other people. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, February 03, 2005 9:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] good god! stop the damn auto-replys! Every freekin' time I post something to this list I get bombarded with out of office auto-replys. Is there no way to stop this? (other than not posting to the list..) -Matthew ___ 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] Broadvoice problems with outbound calls {Scanned}
David,
Here is why register line, not sure if it would be the same effect as yours:
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]/s
- Original Message -
From: David Shaw [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 02, 2005 4:49 PM
Subject: [Asterisk-Users] Broadvoice problems with outbound calls {Scanned}
Hello All, I sign up with $5.99 broadvoice plan. I made in and outbound
calls OK. I upgraded to unlimited world and now I have problems with
outbound calls. I called broadvoice and they said they would get back it
me.
Here are my sip and extension files.
sip.conf
register = XX:[EMAIL PROTECTED]
[broadvoice]
type=friend
username=XX
fromuser=XX
secret=passwd
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=from-broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
allow=ulaw
extensions.conf
exten = _NXX,1,Dial(${TRUNKL2}/${EXTEN})
exten = _NXX,2,Dial(${TRUNKL3}/${EXTEN})
exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _1NXXNXX,3,Dial(${TRUNKL3}/${EXTEN})
exten = _01144XX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _01144XX,2,Dial(${TRUNKL3}/${EXTEN})
Thanks, David
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RE: [Asterisk-Users] howto answer a call in a queue
I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? What's happening on the cli? You should try to start asterisk with asterisk -vdc. Now you should see, what's going on. What kind of phone do you use, perhaps you could use a softclient. SJPhone runs very stable for me. Once more, do it as easy as possible, save your /etc/asterisk/*.* and use only files, you really need. Guido TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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] RE: Cisco 7960G phone crashes during SIP upgrade
I found the phones to be very flaky. Make SURE: 1. The txt files are unix txt files. NO CR/LF at end of line. If you edit the files on a PC use wordpad only. If you use Notepad they will ADD extra characters that causes the phone to reject the files. Do not save them as Unicode text files, this is cause it to fail as well. 2. The 7.x SIP image is not a full image, you have to start with 6.3 or 6.0. I have had success starting with 6.3 on some phones, but others required me to start with 6.0 3. Upgrade path 6.0 6.3 7.3 4. The 7.3 upgrade fails because the directions on how to upgrade are wrong. Try this. a. Edit OS79XX.TXT and put P003-07-3-00 as image name b. Edit SIPDefault.txt and put image name as P003-07-3-00 Reboot the phone. The phone will upgrade the universal loader application, then fail on loading the 7.3 application w/ a Protocol Application Invalid error. DO NOT UNPLUG THE PHONE: c. Now Edit SIPDefault.txt and change image name to P0S3-07-3-00 When the phone reboots by itself, it will upgrade the SIP image to 7.3 and hopefully you are done. I have found with no explanation that sometimes the phone will take multiple reboot with errors before it will work. Cisco says there is a Checklist of things it is looking for. Apparently this checklist progresses from each successive reboot. If you unplug the phone all you are doing is starting over again. Be patient it could take 20 - 30 minutes for each phone. I am not an expert, I have successfully upgraded 20 phones. This was the best I could figure out through trial and error. If you get a checksum error on the SIP 6.3 image, the problem is with your config files not the image. Good luck. I have found upgrading polycoms easier. Found the phone quality to be very good. They have a bug in their 1.3 image that affects phone to phone dialing on the polycom. That appears to have gone away when I upgraded to 1.4 Gene -- Message: 8 Date: Thu, 3 Feb 2005 13:18:28 + From: Nicolas Chabbey [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) 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] Busy Extension Ring to alternate.
I know that with Voicemail you can either do voicemail(uextension) or voicemail(bextension), but with the Sipura SPA-841's I need to be able to roll lines from one extension to an alternate on the phone. For example: If extension 100 is busy, it will ring extension 120 on the same phone, and if that is busy it will ring 140, and then if it hits line 4 with no response, it will then finally go into voicemail. Could I do: exten = 100,1,DIAL(SIP/b100,20,rt) exten = 100,2,DIAL(SIP/b120,20,rt) exten = 100,3,DIAL(SIP/b140,20,rt) exten = 100,n,Voicemail(b100) exten = 100,s+1,Hangup Would this work? Thanks, ~Dan P.S. Has anyone found resources on how to program the LED's on the SPA-841's? ___ 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 7940 [SIP], DTMF and Voicemail
set 'DTMF_inband: 1' in your SIPDefault.cnf to have your voicemail work. Craig - Original Message - From: Derek Conniffe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 5:41 PM Subject: Re: [Asterisk-Users] Cisco 7940 [SIP], DTMF and Voicemail Hi Doug, Thanks very much - sorry I didn't see this in the wiki. The details you quoted worked perfectly and I'm now on firmware V7. I still have the no DTML tones recognised in voicemail but thats another problem. All the best! Derek Doug Lytle wrote: Derek Conniffe wrote: Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). Derek, I found this in the comments field on the wiki, hope this helps on your upgrade, it did for me, I got the same error upgrading from 6.2 to 7.3. It asked for the same files and then got the error invalid protocol whatever. Someone else with these phones gave me the solution. When you upgrade the firmware you need to have the image_version number in 2 files. The os79xx file and the sipdefault file. In all previous upgrades the version was exactly the same example P0S30203 in the OS79XX.txt and image_version P0S30203 in the sipdefault.cnf file. However when upgrading to 7.3 you need 2 different names. in OS79XX.txt you need to write P003-07-3-00 and P0S3-07-3-00 in SIPDefault.cnf then you just need the 4 files in the TFTP server directory, P003-07-3-00.bin, P003-07-3-00.sbn, P0S3-07-3-00.loads, P0S3-07-3-00.sb2. Also if you are upgrading for the 1st time to the inital version P0S30203 and it doesn't work, you make have an error in the status messages section of the phone about a buffer overflow. This can be solved by deleting everything out of sipdefault.cnf except for the image_version: P0S30203 line. Once you have loaded the firmware you can add all the other stuff. This is some buffer overflow error that doesn't let the phone download a file that is over a certain kb in size. Damian ___ 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 -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.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 ___ 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] good god! stop the damn auto-replys!
Matthew Boehm wrote: Every freekin' time I post something to this list I get bombarded with out of office auto-replys. Is there no way to stop this? (other than not posting to the list..) I'd love it if the Mailman software actually handled bounces correctly. Almost every time I post a message to the mailing list I get a message within a day or so, indicating that messages being sent to me have been bouncing. The software even attaches one of the bounces. It's ALWAYS a message that I sent to the mailing list that bounced when it was delivered to someone's mailbox. i.e. Mailman thinks a message that I sent to the list, and bounced when being delivered to a person on the mailing list, is actually a bounce from MY account. Utterly stupid. ___ 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] howto answer a call in a queue
On the CLI everything seems to be ok, the call enters the queue plays the message, on the CLI, appear as a call entering the queue and then show a message saying wich agent is assigned to it! can you send me your config, maybe there is something im doing wrong, thnx for all your help!! Edgar I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? What's happening on the cli? You should try to start asterisk with asterisk -vdc. Now you should see, what's going on. What kind of phone do you use, perhaps you could use a softclient. SJPhone runs very stable for me. Once more, do it as easy as possible, save your /etc/asterisk/*.* and use only files, you really need. Guido TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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
RE: [Asterisk-Users] good god! stop the damn auto-replys!
Oh wow, thanks for clearing that up for me, I didn't understand why I was getting those. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, February 03, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] good god! stop the damn auto-replys! Matthew Boehm wrote: Every freekin' time I post something to this list I get bombarded with out of office auto-replys. Is there no way to stop this? (other than not posting to the list..) I'd love it if the Mailman software actually handled bounces correctly. Almost every time I post a message to the mailing list I get a message within a day or so, indicating that messages being sent to me have been bouncing. The software even attaches one of the bounces. It's ALWAYS a message that I sent to the mailing list that bounced when it was delivered to someone's mailbox. i.e. Mailman thinks a message that I sent to the list, and bounced when being delivered to a person on the mailing list, is actually a bounce from MY account. Utterly stupid. ___ 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] howto answer a call in a queue
im using zultys sw phone, very nice and works very stable!! TIA Edgar I tried everything you said, but its the same thing, when a call enters plays the sound and then is directly connected to one operator, on the operator phone only a beep i heard, what other thing can i try?? What's happening on the cli? You should try to start asterisk with asterisk -vdc. Now you should see, what's going on. What kind of phone do you use, perhaps you could use a softclient. SJPhone runs very stable for me. Once more, do it as easy as possible, save your /etc/asterisk/*.* and use only files, you really need. Guido TIA Edgar My suggestions: Try first the easy (working) configuration then your best solution step by step. comment out leavewhenempty=yes ;it did not work in my system... strategy = ringall ; seems to work don't use groups in the first step ;Play an announcement as the first priority exten = 76522,1,Playback(some_announce) ;even when using an empty file exten = 76522,2,Queue(esculapio|tT|||300) exten = 76522,3,Playback(some_announce_after_leaving_queue) ; if nobody answers the call exten = 76522,4,Hangup I had similiar problem in working with queues. Hope this helps a bit more... Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 3. Februar 2005 09:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your help, here are my config for the queue, agents.conf [agents] musiconhold = random autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez queues.conf [esculapio] leavewhenempty = yes music = random strategy = fewestcalls member = Agent/@1 extensions.conf [ext-acd] exten = 90,1,Answer exten = 90,2,SetMusicOnHold(none) exten = 90,3,Wait,1 exten = 90,4,AgentLogin ;Queue configuration exten = 76522,1,Answer exten = 76522,2,Wait,1 exten = 76522,3,Queue(esculapio|tT|||300) exten = 76522,5,Hangup is my configuration correct?? im using the leavewhenempty = yes option, but when there are no agents the call still enters the queue, thanks for your help TIA Edgar Sometime ago, I wrote an example of a functional queue scenario. Perhaps you give it a try. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue Btw, how is the queue command invoked in your extensions.conf? Post your relevant sections of queues.conf, agents.conf and extensions.conf. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 18:23 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] howto answer a call in a queue Thanks for your answer, i got ackcall=yes but the call when enters only ring once in the agent phone and connect directly, agents.conf [agents] autologoff=15 wrapuptime=5000 ackcall=yes group=1 agent = 1001,3101,Edgar de Leon agent = 1002,,Jorge Cabrera agent = 1003,,Nati del Pozo agent = 1004,,Emilio Perez agent = 1005,,Diego Torres agent = 1006,,Antonio Lopez agent = 1007,,Luis Carlos agent = 1008,,Luis Bonifacio agent = 1009,,Javier Gonzalez what do you think am i doing wrong?? TIA Edgar I think, ackcall=yes should do the job. Guido Hecken -Ursprüngliche Nachricht- Von: Edgar de Leon [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 2. Februar 2005 15:56 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] howto answer a call in a queue hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar ___ 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] Re: problem in compiling asterisk-addons
i am using asterisk-1.0.5 latest available stable version downloaded from www.asterisk.org __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Individual contexts pending on Caller-ID?
Hi! Is it possible to handle incoming calls with different contexts pending on the callerid ? E.g. like you are able to define different contexts on each Zap-channel. 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] Automated CallbackLogin
Hi, we want to provide our users with a Click To Login interface for the AgentCallbackLogin. Any sample.call or AGI anyone has developed out there? Any and all help is greatly appreciated. -- Best regards Peer Oliver Schmidt ___ 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 crashes from time to time
Inline... Far too many variables for anyone to even guess at the root cause. Problem could be related to slight differences in o/s libraries between systems, coding problems within asterisk, etc. You 're right, it could be every thing There were some issues reported with cvs head in January relative to hangs, etc. Are they reported in the bugtracker, or in the mailing list? Not all for sure. If you watch the -cvs and -user list, you'd see folks with seg faults, etc, and not too long after that you see a change come through -cvs. Sometimes with comments like 'fix silly typo', etc. Given that cvs head is actually development, at any point in time there could easily be various problems (expected). To try to recreate historically whether you caught a cvs head version that had errors is almost impossible. That's why its important to run cvs head in some sort of pre-production test environment before promoting the code into a customer's machine, etc. (That implies beating the hell out of your test environment.) Might consider changing /etc/asterisk/logger.conf and add debug to the list. Then after a failure, at least look at /var/log/asterisk/debug messages. Yes, this was the first thing, I did after the crash showed up. I simply forgot to enable it, since this production server ran long time without problems. But now, following murphy's law, the next crash will never happen ;-) For additional info, I'd suggest compiling the code on one of thse machines to see if it complains about missing/inappropriate items. After these machines were setup, we compiled new code on every machine, since we started with an older version of Asterisk in November 2004. The compiling of asterisk did not show me any relevant (?) errors. But I remember there were some statements (Warnings) in the console output of the make process, I didn't understand. Is this output logged in addition to the console in a logfile somewhere? If so, one could examine this output and hopefully get some hints... The only two (key) log methods that I know of is to run the cli with several -'s, and turn on debugging in the logger.conf file (which may require you to config /etc/syslog.conf to catch them). Then look at /var/log/asterisk/debug after a failure. (There are other debug modes, but not sure I'd use those to catch a production problem. The one's I know about are primarily intended for development debugging. Other folks might contribute hints here.) ___ 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] periodic clicking sound
I hear a faint clicking sound (periodic in nature, a click once every 2-3 mins.) on my SIP phone when the call is going through TDM04B. Is there any setting in Zapata that would eliminate it. Any help would be appreciated. Regards, Goutam ___ 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 7960G phone crashes during SIP upgrade
If this approach is not working then there is something else wrong. Is the phone currently running SCCP or SIP? If SIP what version. Nicolas Chabbey wrote: Thanks for your reply. I've already seen the wiki page concerning the xml configuration file problem and mine have absolutly no comments on it, just the minimal lines needed. I've just tried to put the 'P0S30202' on OS79XX and as expected it crash. I think i'll not be able to upgrade to SIP by this way as all name begining with P0S3 or P0M (for MGCP) will automatically hang the device. On Thu, 03 Feb 2005 09:26:56 -0500, Steve Blair [EMAIL PROTECTED] wrote: Nicholas: You need to convert from SCCP to SIP by loading image_version: P0S30202 first. Use the OS79XX.TXT file to specify this version. After that upgrade to each newer release P0S30203, P0S3-03-2-00, etc in the same fashion. Going from 6.3 to 7.0 the loader process changes. You need the OS79XX.TXT file, the SIPmac.cnf and SEPmac.cnf.xml for the phone. From 7.1 on you don't need the OS79XX.TXT file anymore. Nicolas Chabbey wrote: Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) Thanks - Nicolas Chabbey [EMAIL PROTECTED] Leafnet Networking Research Laboratory http://www.bgp6.info - ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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] Re: load balancing 20 asterisk servers
Backhoe's are pretty indiscriminatethey'll cut copper just as easily as fiber. - Original Message - From: David Brodbeck [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 9:32 AM Subject: RE: [Asterisk-Users] Re: load balancing 20 asterisk servers -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] You have fiber-seeking-backhoes in your area? Wow! They're everywhere, man! When I was in college an entire nearby town lost all phone service for 24 hours due to a backhoe cutting a fiber optic cable. 3,000 people with no way of calling emergency services for an entire day. ___ 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] Re: load balancing 20 asterisk servers
Don't confuse load balacing with failover. They are quite different beasts and are handled differently. (sure, they can be combined into one solution, but they are still effectively very different) Round Robin DNS based load balancing is still a viable load balancing solution (read some of the early Netscape white papers compaing round robin DNS to intelligent load balancers, the results were almost identical if not BETTER with round robin DNS.) As far as the failover part, all thats needed is a simple IP Address takeover. nowadays, ip address takeovers is VERY simple. change an ip address, possibly add a proxy arp, and clear the arp cache (or send a garp packet) I'm not necessarially saying this is or should be the solution for this specific problem, i havent really spent any time thinking about SIP. i'm just suggesting a possible easy solution, what people are saying, using SER to redirect to an asterisk server (thats the load balancing piece), and then simple IP Takeover for failover (why buy an expensive cisco box for doing something as easy as ARP) -Mark On Thu, 03 Feb 2005 13:38:49 +0100, Patrick [EMAIL PROTECTED] wrote: Rich Adamson wrote: [snip] I'm not aware of any balancers that can do that today. Afaik Cisco is working on SIP aware loadbalancer functionality. Don't know what the status is and since it's Cisco I'm sure it will cost a bundle. Regards, 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 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] Concurrent calls
Is there any way to quickly poll an asterisk server for concurrent call count? Preferably from like a perl or PHP script. -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
Re: [Asterisk-Users] Echo Problem
Brian M. Arlinghaus wrote: I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On the 7960s, the echo is quite bad. On the TDM ports, it is there, but not as bad. I have tried setting echo cancellation to various numbers, but have had no luck. This began after a HEAD version of * was installed. Since then, I installed what I think is the latest stable version (Asterisk CVS-v1-0-12/14/04-16:49:32) and the echo is still there. A support guy at Digium said it was a SIP problem. Just wanted to second this. I have about 20 7940's, 2 7960, and a 4 port FXS for fax machines going into a Bellsouth T1 (pri) and we get echo on some calls. I turned on echotraining (not for the faxes of course -- echocancelwhenbridged=no) and it will train out, but I thought that voip - pri could not have echo problems. Anyway, please keep me updated if you figure out a (real) solution to this. Jeb -- Jeb Campbell [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] Re: load balancing 20 asterisk servers
Nortel AAS-2000 range of LB's can do this today. Giles - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 12:38 PM Subject: Re: [Asterisk-Users] Re: load balancing 20 asterisk servers Rich Adamson wrote: [snip] I'm not aware of any balancers that can do that today. Afaik Cisco is working on SIP aware loadbalancer functionality. Don't know what the status is and since it's Cisco I'm sure it will cost a bundle. Regards, 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 -- This message has been scanned for viruses and dangerous content by Swift Internet, and is believed to be clean. ___ 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] free pocketPC softphone (toshiba e750)
Hi all I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I didnt found any free softphones for my Toshiba. X lite's versions for pocketPC isnt free :( Did someone used before a free softphone for pocketPC? witch one? Thanks Joao Pereira www.fccn.pt ___ 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 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
I use
pritrustusercid = no
In zapata.conf and then it seems to work.
No idea if it is a bug or not or if this is a proper solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, February 01, 2005 10:11 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
I tried to get callerid working the normal way but the cid is never
passed to the phone.
It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in
extensions.conf
which I found in the wiki:
http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc
Is this intended behaviour, or still a bug?
It does work but it only shows one zero even though I have
nationalprefix = 0 internationalprefix = 00 in zapata.conf
I guess it should show a double zero because there is already a zero
prefix in the SetCIDNum(0${PRI_NETWORK_CID})?
I haven't received any international calls yet but will they not show up
with only one zero now?
Cheers!
Remco
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[Asterisk-Users] Asterisk Dialplan command PPPD released
Hello all! Sirrix AG, Saarbrücken (manufacturer of the Sirrix.PCI4S0 4-port ISDN card for Asterisk) has released the new Asterisk dialplan command PPPD (app_pppd). It allows to connect a Linux PPP daemon to an arbitrary digital (ISDN) Asterisk channel to provide RAS dialin and dialout. The PPPD command has successfully been tested with Sirrix.PCI4S0 cards and a standard ISDN4Linux ipppd on the other side. The PPPD command uses the Linux PPP daemon without additional patches. The PPP daemon must support synchronous PPP mode. PPP support must be enabled in the Kernel. More information about the PPPD command can be found at http://www.voip-info.org/wiki-Asterisk+cmd+PPPD Sirrix AG will be present at CeBIT Hall 9 Booth D09 and show various applications of the Sirrix.PCI4S0 in Asterisk environments. Thank you, Oskar Senft. -- Sirrix AG security technologies - http://www.sirrix-ag.de Oskar Senft eMail: [EMAIL PROTECTED] Tel +49(681)301 409 92Fax +49(681)301 409 91 ___ 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 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
Also just adding
callerid=asreceived
To zapata.conf also seems to work.
Works for local or national calls where I am.
I don't know about international calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, February 01, 2005 10:11 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
I tried to get callerid working the normal way but the cid is never
passed to the phone.
It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in
extensions.conf
which I found in the wiki:
http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc
Is this intended behaviour, or still a bug?
It does work but it only shows one zero even though I have
nationalprefix = 0 internationalprefix = 00 in zapata.conf
I guess it should show a double zero because there is already a zero
prefix in the SetCIDNum(0${PRI_NETWORK_CID})?
I haven't received any international calls yet but will they not show up
with only one zero now?
Cheers!
Remco
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Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
Don't confuse load balacing with failover. They are quite different beasts and are handled differently. (sure, they can be combined into one solution, but they are still effectively very different) Round Robin DNS based load balancing is still a viable load balancing solution (read some of the early Netscape white papers compaing round robin DNS to intelligent load balancers, the results were almost identical if not BETTER with round robin DNS.) That's true for well behaved systems, but isn't true with the majority of sip phones and adapters. The majority do not pay any attention to the ttl fields in the dns response, and thus won't try another dns query during follow on attempts. Plus, a fair number of phones and adapters have problems with dns and the work around (as suggested by the OEM) is to use hard coded IP addresses. There are actually far more exception cases then there are solid sip devices (from a dns perspective), therefore round robin with sip boxes will not see the same results as the white paper you've referred to. I have not tried to prove this, but I'd bet at least some money that sip phones and adapters don't do a dns lookup each time a call is placed. It _might_ do a lookup on each register, but not each call. As far as the failover part, all thats needed is a simple IP Address takeover. nowadays, ip address takeovers is VERY simple. change an ip address, possibly add a proxy arp, and clear the arp cache (or send a garp packet) The OP was asking about load balancing 20 servers. IP address takeover wouldn't apply nor would it scale. I'm not necessarially saying this is or should be the solution for this specific problem, i havent really spent any time thinking about SIP. i'm just suggesting a possible easy solution, what people are saying, using SER to redirect to an asterisk server (thats the load balancing piece), and then simple IP Takeover for failover (why buy an expensive cisco box for doing something as easy as ARP) That does sound like a very reasonable approach. :) ___ 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] Different rings
How can I get a different ring tone when I get a call from the PSTN on my Cisco IP Phone 7960? I want one ring tone when it's an internal call (coming from another SIP extension on my network) and another one when it's coming from the PSTN. I'm using TDM04B cards if that make any difference to answer PSTN calls. Thanks Martin ___ 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: Cisco 7960G phone crashes during SIP upgrade
I can confirm that the Cisco instruction for installing/upgrading to the 7.3 SIP image do not work. When I originally installed SIP on some brand new phones Cisco TAC indicated that a phone has to run 6.3 before it can be upgraded to 7.3. Loading the 6.3 SIP image has been a success. I've had no problems switching between 6.3 SIP and various SCCP images back and forth. I might try your suggested method to get up to the 7.3 SIP image. However, it doesn't sound like a hassle free method of loading SIP at all. I'd much prefer if I could tell the phone via the dhcp server to load from tftp server x.x.x.x and automatically loads SIP or load from tftp server y.y.y.y and automatically load SCCP (where x.x.x.x is Asterisk and y.y.y.y is Cisco CCM). Adi On Thu, 3 Feb 2005, Gene Willingham wrote: I found the phones to be very flaky. Make SURE: 1. The txt files are unix txt files. NO CR/LF at end of line. If you edit the files on a PC use wordpad only. If you use Notepad they will ADD extra characters that causes the phone to reject the files. Do not save them as Unicode text files, this is cause it to fail as well. 2. The 7.x SIP image is not a full image, you have to start with 6.3 or 6.0. I have had success starting with 6.3 on some phones, but others required me to start with 6.0 3. Upgrade path 6.0 6.3 7.3 4. The 7.3 upgrade fails because the directions on how to upgrade are wrong. Try this. a. Edit OS79XX.TXT and put P003-07-3-00 as image name b. Edit SIPDefault.txt and put image name as P003-07-3-00 Reboot the phone. The phone will upgrade the universal loader application, then fail on loading the 7.3 application w/ a Protocol Application Invalid error. DO NOT UNPLUG THE PHONE: c. Now Edit SIPDefault.txt and change image name to P0S3-07-3-00 When the phone reboots by itself, it will upgrade the SIP image to 7.3 and hopefully you are done. I have found with no explanation that sometimes the phone will take multiple reboot with errors before it will work. Cisco says there is a Checklist of things it is looking for. Apparently this checklist progresses from each successive reboot. If you unplug the phone all you are doing is starting over again. Be patient it could take 20 - 30 minutes for each phone. I am not an expert, I have successfully upgraded 20 phones. This was the best I could figure out through trial and error. If you get a checksum error on the SIP 6.3 image, the problem is with your config files not the image. Good luck. I have found upgrading polycoms easier. Found the phone quality to be very good. They have a bug in their 1.3 image that affects phone to phone dialing on the polycom. That appears to have gone away when I upgraded to 1.4 Gene -- Message: 8 Date: Thu, 3 Feb 2005 13:18:28 + From: Nicolas Chabbey [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) 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
Re: [Asterisk-Users] Re: load balancing 20 asterisk servers
Mark, I've been following this thread with some interest as we're gearing up for load/failover processing. Can you elaborate on the garp and IP takeover process, like what software packages do that in Linux or point me to a site for more info? Thanks, Jon - Original Message - From: Mark Musone [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 10:36 AM Subject: Re: [Asterisk-Users] Re: load balancing 20 asterisk servers Don't confuse load balacing with failover. They are quite different beasts and are handled differently. (sure, they can be combined into one solution, but they are still effectively very different) Round Robin DNS based load balancing is still a viable load balancing solution (read some of the early Netscape white papers compaing round robin DNS to intelligent load balancers, the results were almost identical if not BETTER with round robin DNS.) As far as the failover part, all thats needed is a simple IP Address takeover. nowadays, ip address takeovers is VERY simple. change an ip address, possibly add a proxy arp, and clear the arp cache (or send a garp packet) I'm not necessarially saying this is or should be the solution for this specific problem, i havent really spent any time thinking about SIP. i'm just suggesting a possible easy solution, what people are saying, using SER to redirect to an asterisk server (thats the load balancing piece), and then simple IP Takeover for failover (why buy an expensive cisco box for doing something as easy as ARP) -Mark On Thu, 03 Feb 2005 13:38:49 +0100, Patrick [EMAIL PROTECTED] wrote: Rich Adamson wrote: [snip] I'm not aware of any balancers that can do that today. Afaik Cisco is working on SIP aware loadbalancer functionality. Don't know what the status is and since it's Cisco I'm sure it will cost a bundle. Regards, 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 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] Echo Problem
I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On the 7960s, the echo is quite bad. On the TDM ports, it is there, but not as bad. I have tried setting echo cancellation to various numbers, but have had no luck. This began after a HEAD version of * was installed. Since then, I installed what I think is the latest stable version (Asterisk CVS-v1-0-12/14/04-16:49:32) and the echo is still there. A support guy at Digium said it was a SIP problem. Just wanted to second this. I have about 20 7940's, 2 7960, and a 4 port FXS for fax machines going into a Bellsouth T1 (pri) and we get echo on some calls. I turned on echotraining (not for the faxes of course -- echocancelwhenbridged=no) and it will train out, but I thought that voip - pri could not have echo problems. Anyway, please keep me updated if you figure out a (real) solution to this. echotraining=800 did fix the OP's problem. But, there can still be far-end echo even with PRI's. Those cases involve hybrid issues at some distant end that are difficult at best to address at your end. As has been stated many times before, the echo canceller within * is not as good/reliable as commercial can's and won't handle some of the far-end echo problems. ___ 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 with SourdCard
Hello , You don`t need a sound card. Only you will need a sound card if you have a Softphone. Regards. Ramon Jesus Garcia. Giovanni Miano [EMAIL PROTECTED] Enviado por: [EMAIL PROTECTED] 02/02/2005 18:49 Por favor, responda a Giovanni Miano [EMAIL PROTECTED]; Por favor, responda a Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para Asterisk-Users@lists.digium.com cc Asunto [Asterisk-Users] Asterisk with SourdCard My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ? ___ 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] Call Forward Loop
I'm not sure if this was discussed before, but can someone tell me if there is a solution for a problem when two (for example SIP) phones are unconditionally forwarded to each other. In my previous attempts this situation usually killed asterisk. Any comments and suggestions would be appreciated. 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
Re: [Asterisk-Users] Re: problem in compiling asterisk-addons
*bamn* damn..hate it when i'm right. you are attempting to use asterisk-1.0.5 with asterisk-addons-CVS. insert *VERY LOUD buzzer* pick 1 version and use that version in both cases. you are not going to be able to use res_config_mysql with 1.0.5 anyway as RealTime is a CVS-0nly deal. -Matthew - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 03, 2005 9:10 AM Subject: Re: [Asterisk-Users] Re: problem in compiling asterisk-addons i am using asterisk-1.0.5 latest available stable version downloaded from www.asterisk.org __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] good god! stop the damn auto-replys!
On Thu, 2005-02-03 at 09:48 -0500, dean collins wrote: Yep, send the out of office replies various junk files :) Everyone yesterday got some nice 4 mb photos of the snow in New York that I took last week. It's just a gentle reminder that they should learn how to use technology before they implement it against other people. I have been resorting to putting a rule in my sieve scripts to bounce those messages to another email address at the same domain it came from. Some time back when we had the guy who no longer worked for whatever employer (ploviar?), I found all their sales addresses and bounced that message to each one. I think it might have stopped pretty quickly after that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, February 03, 2005 9:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] good god! stop the damn auto-replys! Every freekin' time I post something to this list I get bombarded with out of office auto-replys. Is there no way to stop this? (other than not posting to the list..) -Matthew -- Steven Critchfield [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] Chan_Capi initial deadlock
On Thursday, 03 February, 2005 13:54 : Felix Deierlein [EMAIL PROTECTED] wrote: I had applied the patch and it got much better. Now I only have problems every two days eb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries! Feb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries! Feb 3 14:04:12 WARNING[23065]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'CAPI[contr1/1429092]/279', 10 retries! Any idea? When you have this message, attach gdb to asterisk and type : thread apply all bt in the list, select the lastest asterisk function, just before libc, and type frame (this number). Example : (gdb) thread apply all bt Thread 13 (Thread 19469 (LWP 3466)): #0 0x4016bde1 in nanosleep () from /lib/libc.so.6 #1 0x40195e8e in usleep () from /lib/libc.so.6 #2 0x402af5f2 in capi_activehangup (c=0x40508ba8) at chan_capi.c:563 #3 0x402af7c7 in capi_hangup (c=0x40508ba8) at chan_capi.c:606 #4 0x0805945c in ast_hangup (chan=0x40508ba8) at channel.c:741 #5 0x08072b7f in ast_pbx_run (c=0x40508ba8) at pbx.c:1968 #6 0x08079036 in pbx_thread (data=0x40508ba8) at pbx.c:1980 #7 0x400200ba in pthread_start_thread () from /lib/libpthread.so.0 (gdb) frame 2 #2 0x402af5f2 in capi_activehangup (c=0x40508ba8) at chan_capi.c:563 563 usleep(1); And past the results of these commands. Good luck -- Carl ___ 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] Difference between Asterisk and VOCAL
Hi All, I am new to Asterisk and been searching Internet for long about VoIP. I am quite confuse about Asterisk and VOCAL since Asterisk can also support SIP. Can anyone show some lights? Cheers, Stephen Liew begin:vcard fn:Stephen Liew n:Liew;Stephen adr:Tmn Melodies;;14A, Jln Geronggang;Johor Bahru;Johor;80250;Malaysia email;internet:[EMAIL PROTECTED] tel;work:+(60) 7 334 9781 tel;fax:+(60) 7 334 5502 tel;cell:+(60)-12-7107350 url:http://www.revoltel.com version:2.1 end:vcard ___ 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] Individual contexts pending on Caller-ID?
Daniel Nyström wrote: Hi! Is it possible to handle incoming calls with different contexts pending on the callerid ? E.g. like you are able to define different contexts on each Zap-channel. Just dump all the calls to a sorter context, and build your rules there. Either type in all the relavent telephone numbers, or use a database lookup tool. The last command ran here would be: Goto(VARIABLE_HOLDING_CONTEXT_NAME, 1) -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.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] IAXy Hung, Power-cycle Required
Has anyone had good success with the IAXy? I've tried everything including PAT on the IAX2 port to the IAXy device to no avail (using the alternate server parameter). I guess a call to Digium is in order! Regards, --- Gavin Adams Promisant (USA) Inc. O: 770-913-3727 F: 770-913-3726 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryan Field-Elliot Sent: Tuesday, February 01, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAXy Hung, Power-cycle Required On Wed, 2005-01-26 at 15:59 -0500, Paul Dugas wrote: I've got a single IAXy installed in a little office nearby and got a call from someone on site a finew mintues ago. Apparently they couldn't make a call on that extension. They'd pick up the phone and get nothing; no dial-tone. Has snyone else had trouble with these things sticking like this? Paul Yes - we are having the exact same problem with a portion of our IAXys in the field. In all cases the IAXys are behind simple SOHO firewalls like the Linksys. After an idle period - perhaps 1-3 days - they just stop working, in both directions, but a simple power cycle restores functionality. We have an open support incident with Digium but have not yet heard back. FWIW we have stopped selling deploying the IAXys until we have a resolution to the problem. Bryan ___ 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: load balancing 20 asterisk servers
On Feb 3, 2005, at 17:08, Jon Bebeau wrote: Mark, I've been following this thread with some interest as we're gearing up for load/failover processing. Can you elaborate on the garp and IP takeover process, like what software packages do that in Linux or point me to a site for more info? http://www.linux-ha.org/ jens ___ 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: Cisco 7960G phone crashes during SIP upgrade
The 7.3 zip file contains the wrong filenames from their website. If you watch the status messages, you should see a incorrect loads or invalid loads flash before the phone continually reboots. I cannot remember exactly which files I renamed to make it work, but I edited the .loads file, and renamed sbn or bin, etc and finally got it all working to 7.3 from 6.5 Skinny image. -Greg Adi Linden wrote: I can confirm that the Cisco instruction for installing/upgrading to the 7.3 SIP image do not work. When I originally installed SIP on some brand new phones Cisco TAC indicated that a phone has to run 6.3 before it can be upgraded to 7.3. Loading the 6.3 SIP image has been a success. I've had no problems switching between 6.3 SIP and various SCCP images back and forth. I might try your suggested method to get up to the 7.3 SIP image. However, it doesn't sound like a hassle free method of loading SIP at all. I'd much prefer if I could tell the phone via the dhcp server to load from tftp server x.x.x.x and automatically loads SIP or load from tftp server y.y.y.y and automatically load SCCP (where x.x.x.x is Asterisk and y.y.y.y is Cisco CCM). Adi On Thu, 3 Feb 2005, Gene Willingham wrote: I found the phones to be very flaky. Make SURE: 1. The txt files are unix txt files. NO CR/LF at end of line. If you edit the files on a PC use wordpad only. If you use Notepad they will ADD extra characters that causes the phone to reject the files. Do not save them as Unicode text files, this is cause it to fail as well. 2. The 7.x SIP image is not a full image, you have to start with 6.3 or 6.0. I have had success starting with 6.3 on some phones, but others required me to start with 6.0 3. Upgrade path 6.0 6.3 7.3 4. The 7.3 upgrade fails because the directions on how to upgrade are wrong. Try this. a. Edit OS79XX.TXT and put P003-07-3-00 as image name b. Edit SIPDefault.txt and put image name as P003-07-3-00 Reboot the phone. The phone will upgrade the universal loader application, then fail on loading the 7.3 application w/ a Protocol Application Invalid error. DO NOT UNPLUG THE PHONE: c. Now Edit SIPDefault.txt and change image name to P0S3-07-3-00 When the phone reboots by itself, it will upgrade the SIP image to 7.3 and hopefully you are done. I have found with no explanation that sometimes the phone will take multiple reboot with errors before it will work. Cisco says there is a Checklist of things it is looking for. Apparently this checklist progresses from each successive reboot. If you unplug the phone all you are doing is starting over again. Be patient it could take 20 - 30 minutes for each phone. I am not an expert, I have successfully upgraded 20 phones. This was the best I could figure out through trial and error. If you get a checksum error on the SIP 6.3 image, the problem is with your config files not the image. Good luck. I have found upgrading polycoms easier. Found the phone quality to be very good. They have a bug in their 1.3 image that affects phone to phone dialing on the polycom. That appears to have gone away when I upgraded to 1.4 Gene -- Message: 8 Date: Thu, 3 Feb 2005 13:18:28 + From: Nicolas Chabbey [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960G phone crashes during SIP upgrade To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image file at any moment. If i'm putting a SCCP image file name (without ext.) on the OS79XX.TXT (begining with P003), the phone doesn't crash and request the respective SEPmac.xml file. Unfortunately (again), just after downloading the xml configuration it hang and restart. I've checked the syntax and they's no error on it, if they's one the phone output the error on the display without crashing. Note that i've both put with and without the load information statement, with the same result. Both statical and DHCP configuration has been tried. Maybe it's an hardware failure or i've miss somethings realy important :) 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:
[Asterisk-Users] Re: load balancing 20 asterisk servers
As being said, the cost of HW based solutions is in many cases too expensive to be practical, leave alone to have a spare one to give you true high-availibility. If you complain about DNS caching and timeouts not being respected, you can do a fairly easy thing. As said in the follow-up, you can drop in a data center many tiny boxes. When one box stops working or crashes you can bring down it's interface and give a spare box the same IP of the [now] defunct machine. If the box becomes irresponsive (a real OS crash) and you have IMPI 2.0 capable MoBo's (all intel server boards have that, also some other brands), you can remotely shutdown or reset the machine to avoid IP clashes. Intel has a command line utility (also the graphical console) to manage that, i've using for a while this and is absolutely wonderful. Obiously, you will need a separate LAN with privates IP's to make much of the administration and the DB access, and use the public LAN only for internet-related things, so each box has it's own fixed private IP and only the public IP changes. Indeed, with IMPI 2.0 is possible to remotely power-up a machine (if you are enviromentally concerned... or if the datacenter metters the electricity you use), so you don't need to be running all your spare servers waiting for a failure, maybe only one and have the others shut-down until needed. --- [EMAIL PROTECTED] wrote: Message: 5 Date: Wed, 2 Feb 2005 21:57:42 -0600 (CST) From: Joe Greco [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: load balancing 20 asterisk servers To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I'm trying to stay away from a software based load balancer cause what happens if that server fails? Its far less likely for a piece of dedicated hardware to fail than an actual computer. You really ought to open up one of those pieces of dedicated hardware sometime and see what's inside. Yep, it's software based. Heck, many of the so-called pieces of dedicated hardware are in fact nothing more than a fancy rack mount PC. Open up something like a CacheFlow server and you find an Intel server motherboard, some propietary software, and that is about it. Heck, go on eBay and pick yourself up some of those nice F5 BigIP ... rack mount PC's. Some of the newer stuff is software based with some ASIC assistance for SSL/compression. I know that F5 has made an effort to not look like a PC anymore, for example, and has integrated some switchlike capabilities in their product. Still, when it comes right down to it, the traffic direction logic in these things is software based. Incidentally: one of the /down/sides to these devices, aside from being hellishly expensive, is that when it blows at 5:01PM on a Thursday evening when Friday is Christmas, even if you have the best service contract, it can be a trying experience to get service. PC's have the distinct advantage that you can actually plan to have spare parts available, and on top of it, you can actually build high quality redundant equipment fairly inexpensively. AIC RMC2N-XP Chassis$150 EMACS R2G-6350P Power $300 SuperMicro P4SC8$300 Intel P4-3.0 Prescott $175 Memory as desired CF Adapter $ 20 1GB CompactFlash Boot $ 60 $1005 Toss in a monster passive heatsink and you have a system that isn't particularly susceptible to the loss of any single moving part. Of course, you have to be able to sysadmin your way out of a cardboard box, so it's not like it's cost-free, but here's the thing: If my hypothetical load balancer fails at 5:01PM on Xmas eve, I can: 1) Grab the cold spare I built because it's cheaper to do two of these than a single expensive HW based solution 2) Configure the hot spare I built into production (again because it's cheaper). 3) Grab a desktop PC and stick a few Intel GigE NIC's in it and go to town. 4) At least have a reasonable chance of figuring out some other way to fix things temporarily. So. What's really interesting is that even some networking hardware is actually just computing gear on steroids. I recently saw a SMC 8624T 24-port gigE switch, and it appears to be a bunch of Broadcom GigE chips with a CPU that runs some (can't recall which) embedded OS. VxWorks? ... JG = Miguel Ruiz Velasco Version: OpenKeyServer v1.2 Comment: Extracted from belgium.keyserver.net Signature: 0x59831109 __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.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 crashes from time to time
(There are other debug modes, but not sure I'd use those to catch a production problem. The one's I know about are primarily intended for development debugging. Other folks might contribute hints here.) This reeks of a deadlock, http://voip-info.org/wiki-Asterisk+deadlock see this HowTo Debug a DeadLock in Asterisk i wrote up eons ago on the wiki http://voip-info.org/wiki-Asterisk+debugging ___ 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] IAXy Hung, Power-cycle Required
On Feb 3, 2005, at 17:37, Adams, Gavin-ML wrote: Has anyone had good success with the IAXy? I've tried everything including PAT on the IAX2 port to the IAXy device to no avail (using the alternate server parameter). I guess a call to Digium is in order! Works for me[TM], without fail jens ___ 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] astcc digit timeout
does anyone know how to change the timeout on digit entry in astcc. if you call the app and start entering a pin, you have about 2 seconds to enter the next number or you get timed out. i cannot find any info on this from the lists or google. Thanks, Steve ___ 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
