[asterisk-users] Timer B in sip.conf cannot be changed
I am using 1.8. I need to change timerb to 6500, that is, if there is no response of some sort in 6.5 seconds, consider the call failed and try another route. It does not matter what do I set for the other timers: T1min=100 timert1=100 Timerb=6500 The command "sip show settings" always shows Timer B=32000. Any ideas how can I reduce Timer B? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI Callerid and transfer problem
Hi, We need some help.
We are unable to transfer the incoming call from DAHDI to another
number. We are able to receive calls and dial out fine, but what we
really want to achieve is to transfer the call so that PRI will be
free and also the transferred number will get the received callerid.
Here is dialplan.
[default]
exten => 1999001,1,myprint(${CALLERID(num)})
exten => 1999001,n,transfer(DAHDI/1/14085551234)
exten => 1999001,n,myprint(${TRANSFERSTATUS})
The problem is transfer fails and TRANSFERSTATUS is set to
UNSUPPORTED. It works if we change 'transfer' command to 'Dial' but in
this case it does not pass callerid. facilityenable and transfer is
set to 'yes' in chan_dahdi.conf.
Any hints what we are doing wrong?
Thanks
Jim
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Re: [asterisk-users] Firewall Issue
Hi, (1) Since a few days, I am seeing unexpected (unwanted) calls reaching my asterisk server. Please see attached log files. (2) I believe the source IP of these calls is the IP mentioned under the CHANNELS column. (3) But as per my firewall, these calls should not have reached Asterisk. The should have been dropped by the Firewall. Please suggest if my thinking is in the correct direction, and what should be my next step. Best regards, Sans +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ | calldate| clid | src | dst | dcontext | channel | dstchannel | lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | uniqueid| userfield | dnid| +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ | 2011-08-04 11:23:15 | "000441913561021" | asterisk | s | from-trunk | SIP/94.247.178.106-0285 || Hangup | | 19 | 19 | ANSWERED|3 | | 1312471395.2207 | | 000441913561021 | +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ +-+--+--+-++++-+--+--+-+-+--+-+-+---+-+ | 2011-08-04 15:26:19 | "001441913561025" | asterisk | s | from-trunk | SIP/72.32.198.159-0401 || Hangup | | 18 | 18 | ANSWERED|3 | | 1312485979.6667 | | 001441913561025 | +-+--+--+-++++-+--+--+-+-+--+-+-+---+-+ +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ | 2011-08-04 17:51:12 | "002441913561017" | asterisk | s | from-trunk | SIP/50.28.9.55-04b4 || Hangup | | 19 | 18 | ANSWERED|3 | | 1312494672.7195 | | 002441913561017 | +-+--+--+-++-++-+--+--+-+-+--+-+-+---+-+ +-++--+-++-++-+--+--+-+-+--+-+-+---+---+ | 2011-08-04 16:20:20 | "2441913561035" | asterisk | s | from-trunk | SIP/75.125.193.162-0446 || Hangup | | 16 | 16 | ANSWERED|3 | | 1312489220.6866 | | 2441913561035 | +-++--+-++-++-+--+--+-+-+--+-+-+---+---+ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
Hi, Could you attach iptables-save output. 2011/8/8 RSCL Mumbai > Hi, > > (1) Since a few days, I am seeing unexpected (unwanted) calls reaching my > asterisk server. > Please see attached log files. > > (2) I believe the source IP of these calls is the IP mentioned under the > CHANNELS column. > > (3) But as per my firewall, these calls should not have reached Asterisk. > The should have been dropped by the Firewall. > > > Please suggest if my thinking is in the correct direction, and what should > be my next step. > > Best regards, > Sans > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Dialplan
Use extensions_custom.conf file to update your custom configurations. On Sun, Aug 7, 2011 at 3:59 AM, Barry L. Kline wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > On 08/05/2011 04:32 AM, Richard Zulu wrote: > > > I would like to import my dialplan into freepbx+asterisk since I am > > switching to that...how can I create my own custom dialplan in > > freepbx? > > I'm not sure why you'd want to... freepbx is anathema to custom > dialplans. That said, I believe you end up naming your > "extensions.conf" file to "extensions_additional.conf" and freepbx will > pick it up when it starts. > > It's been a long, long time since I've dealt with freepbx -- in fact I > went the other way: from freepbx+asterisk to pure asterisk. When I was > using freepbx that was the solution you seek. > > Barry > > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.5 (GNU/Linux) > > iD8DBQFOPcAxCFu3bIiwtTARAkjKAKCPCgcoaRyPNs7BXhge7xxcy7C2qQCdF6hx > 2Bwz/YEUSbKFsfzD9V0xX6Q= > =W2Dn > -END PGP SIGNATURE- > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and auto answer
Warren, Thanks, I ended up doing that but it didn't change a thing. I mean, the originating phone does not drop into a conference obviously, but the ringing still goes on and on for 30 secs (my timeout). Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, August 08, 2011 2:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and auto answer On Sun, Aug 7, 2011 at 9:32 PM, Mike wrote: Hi, [paging] exten => s,1,Verbose(1,paging) exten => s,n,Set(TIMEOUT(absolute)=30) ;to prevent call from being stuck exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) exten => s,n,Page(SIP/sipphone) Try changing the Page() to a Dial() command and see if that makes a difference. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail to correctly build 1.8.5 ??
Hi Do you see the loaded modules when not using the conf file, somethinglike this ? cdr_mysql.so MySQL CDR Backend0 res_config_mysql.soMySQL RealTime Configuration Driver 0 app_mysql.so Simple Mysql Interface 0 soeren -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norbert Zawodsky Sent: Sunday, August 07, 2011 8:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fail to correctly build 1.8.5 ?? Hello everybody, I've been using asterisk 1.2 for quite a long time now, but I thought it's time to try a newer version of asterisk. So I downloaded 1.8.5, extracted the tar, ran configure, make, make install ... Everything looks fine (no obvious compile/link errors). But as soon as I start asterisk, it dies with a segfault. I executed asterisk within strace and last action before the segfault was reading cdr_mysql.conf in. I deleted cdr_mysql.conf and asterisk starts normally. I copied cdr_mysql.conf.sample to /etc/asterisk and asterisk starts normally Then I just uncommented the "hostname=" line and asterisk segfaults during startup. So what could be wrong here? I have packages "libmysqlclient-devel" and "libmysqlclient16" installed, both version 5.1.46-2.18 OS is a SuSE 11.3, 64 bit Any hints? Norbert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
Also you can set allowguest=no in sip.conf, if you didn't do it already On 2011.08.08 13:24, RSCL Mumbai wrote: > Hi, > > (1) Since a few days, I am seeing unexpected (unwanted) calls reaching > my asterisk server. > Please see attached log files. > > (2) I believe the source IP of these calls is the IP mentioned under > the CHANNELS column. > > (3) But as per my firewall, these calls should not have reached > Asterisk. The should have been dropped by the Firewall. > > > Please suggest if my thinking is in the correct direction, and what > should be my next step. > > Best regards, > Sans > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
On Mon, Aug 8, 2011 at 4:20 PM, Антон Квашёнкин wrote: > Hi, > > Could you attach iptables-save output. > "iptables-save" output is blank -- no output. Not sure why ? Thx Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
On Mon, Aug 8, 2011 at 5:09 PM, Henrik wrote: > ** > Also you can set allowguest=no in sip.conf, if you didn't do it already > > I will check sip.conf, but logically, the packets should not be reaching Asterisk. IP Tables should have blocked them. Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
For some unknown reason, the firewall script was not executed. Now I get the output of iptables-save. May be this is the reason why unwanted packets hit the system... a big blunder. Sans On Mon, Aug 8, 2011 at 5:44 PM, RSCL Mumbai wrote: > > > On Mon, Aug 8, 2011 at 4:20 PM, Антон Квашёнкин > wrote: > >> Hi, >> >> Could you attach iptables-save output. >> > > "iptables-save" output is blank -- no output. > Not sure why ? > > Thx > Sans > [root@e1 ~]# iptables-save # Generated by iptables-save v1.3.5 on Mon Aug 8 08:19:37 2011 *filter :INPUT DROP [1:78] :FORWARD DROP [0:0] :OUTPUT ACCEPT [2496:492015] -A INPUT -i lo -j ACCEPT -A INPUT -p icmp -m icmp --icmp-type 8 -m state --state NEW -j ACCEPT -A INPUT -i eth1 -m state --state RELATED,ESTABLISHED -j ACCEPT -A INPUT -i eth0 -m state --state RELATED,ESTABLISHED -j ACCEPT -A INPUT -p tcp -m tcp --dport 3100 -j ACCEPT -A INPUT -p tcp -m tcp --dport 4142 -j ACCEPT -A INPUT -i eth0 -p tcp -m tcp --dport 4445 -j ACCEPT -A INPUT -i eth1 -p tcp -m tcp --dport 4445 -j ACCEPT -A INPUT -s 67.18.110.210 -i eth1 -p tcp -m tcp --dport 3306 -j ACCEPT -A INPUT -s 192.168.1.0/255.255.255.0 -i eth0 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 192.168.1.0/255.255.255.0 -i eth0 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 192.168.1.0/255.255.255.0 -i eth0 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 61.16.181.9 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 61.16.181.9 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 61.16.181.9 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 203.109.120.65 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 203.109.120.65 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 203.109.120.65 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.82.128/255.255.255.192 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.82.128/255.255.255.192 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.82.128/255.255.255.192 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.83.0/255.255.255.192 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.83.0/255.255.255.192 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.83.0/255.255.255.192 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.84.0/255.255.255.0 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.84.0/255.255.255.0 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.84.0/255.255.255.0 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 81.201.86.0/255.255.255.192 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.86.0/255.255.255.192 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.201.86.0/255.255.255.192 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 74.55.98.122 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.122 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.122 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 74.55.98.120 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.120 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 74.55.98.120 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 64.154.41.150 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.150 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.150 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 64.154.41.100 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.100 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 64.154.41.100 -i eth1 -p udp -m udp --dport 1:2 -j ACCEPT -A INPUT -s 46.19.209.8/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.209.8/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.209.72/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.209.72/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.8/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.8/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.72/255.255.255.248 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.210.72/255.255.255.248 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.85.224.40/255.255.255.254 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 81.85.224.40/255.255.255.254 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 212.150.88.20/255.255.255.252 -i eth1 -p udp -m udp --dport 5060:5062 -j ACCEPT -A INPUT -s 212.150.88.20/255.255.255.252 -i eth1 -p tcp -m tcp --dport 5060:5062 -j ACCEPT -A INPUT -s 46.19.209.0/255.255.255.128 -i
Re: [asterisk-users] Firewall Issue
lsmod | grep ipt And what distribution do you use? 2011/8/8 RSCL Mumbai > > > On Mon, Aug 8, 2011 at 4:20 PM, Антон Квашёнкин > wrote: > >> Hi, >> >> Could you attach iptables-save output. >> > > "iptables-save" output is blank -- no output. > Not sure why ? > > Thx > Sans > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
Certainly YES, but how you run this script: rc.local, iptables-restore < /etc/iptables or service iptables 2011/8/8 RSCL Mumbai > For some unknown reason, the firewall script was not executed. > Now I get the output of iptables-save. > > May be this is the reason why unwanted packets hit the system... a big > blunder. > > Sans > > > > > > > On Mon, Aug 8, 2011 at 5:44 PM, RSCL Mumbai wrote: > >> >> >> On Mon, Aug 8, 2011 at 4:20 PM, Антон Квашёнкин >> wrote: >> >>> Hi, >>> >>> Could you attach iptables-save output. >>> >> >> "iptables-save" output is blank -- no output. >> Not sure why ? >> >> Thx >> Sans >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
If you take a bit deep analyses on SIP packet you will be able to understand the issue, Iptables filter on layer-3 while SIP is on layer-7. It is easily possible to generate a SIP packet with different source-ip than physical interface. You can also simulate it if you set external-ip=some-else-ip in SIP.com in asterisk. All you SIP packets will contain new some-else-ip while layer-3 headers will still have actual physical interface IP. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Monday, August 08, 2011 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Firewall Issue On Mon, Aug 8, 2011 at 5:09 PM, Henrik wrote: Also you can set allowguest=no in sip.conf, if you didn't do it already I will check sip.conf, but logically, the packets should not be reaching Asterisk. IP Tables should have blocked them. Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail to correctly build 1.8.5 ??
Am 08.08.2011 13:38, schrieb Soeren Malchow (MCon): -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norbert Zawodsky Sent: Sunday, August 07, 2011 8:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fail to correctly build 1.8.5 ?? Hello everybody, I've been using asterisk 1.2 for quite a long time now, but I thought it's time to try a newer version of asterisk. So I downloaded 1.8.5, extracted the tar, ran configure, make, make install ... Everything looks fine (no obvious compile/link errors). But as soon as I start asterisk, it dies with a segfault. I executed asterisk within strace and last action before the segfault was reading cdr_mysql.conf in. I deleted cdr_mysql.conf and asterisk starts normally. I copied cdr_mysql.conf.sample to /etc/asterisk and asterisk starts normally Then I just uncommented the "hostname=" line and asterisk segfaults during startup. So what could be wrong here? I have packages "libmysqlclient-devel" and "libmysqlclient16" installed, both version 5.1.46-2.18 OS is a SuSE 11.3, 64 bit Any hints? Norbert -- Hi Do you see the loaded modules when not using the conf file, somethinglike this ? cdr_mysql.so MySQL CDR Backend0 res_config_mysql.soMySQL RealTime Configuration Driver 0 app_mysql.so Simple Mysql Interface 0 soeren Hi Soeren, Yes, "module show" shows all three of them. norbert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Version 1.8 strange expression error
This expression that worked fine in 1.6.2 is returning an error:
exten =>_X.,n,Set(i=$[${i} + 1])
It needs to add 1 to the value if "i". Did I miss something?
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Re: [asterisk-users] Version 1.8 strange expression error
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of CDR
> Sent: Monday, August 08, 2011 9:42 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Version 1.8 strange expression error
>
> This expression that worked fine in 1.6.2 is returning an error:
>
> exten =>_X.,n,Set(i=$[${i} + 1])
>
> It needs to add 1 to the value if "i". Did I miss something?
Showing us the actual error message might be helpful. Also, add a exten
=>_X.,n,Noop(i is ${i}) and show us the CLI output.
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[asterisk-users] CEL and MySQL
Is anyone using CEL with a MySQL backed at all? I've found a table schema but I'm guessing I need some sort of cel_mysql.conf and don't even have a sample for that. Can anyone give me any pointers as to what files I need to change to get this logging to my MySQL table? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
2011/8/8 Антон Квашёнкин > lsmod | grep ipt > And what distribution do you use? > > [root@e1 ~]# lsmod | grep ipt ipt_REJECT 38977 1 iptable_filter 36161 1 iptable_nat40773 0 ip_nat 53101 1 iptable_nat ip_conntrack 91621 3 xt_state,iptable_nat,ip_nat ip_tables 55201 2 iptable_filter,iptable_nat x_tables 50505 5 ipt_REJECT,xt_tcpudp,xt_state,iptable_nat,ip_tables -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
On Mon, Aug 8, 2011 at 6:03 PM, Faisal Hanif wrote: > If you take a bit deep analyses on SIP packet you will be able to > understand the issue, > > ** ** > > Iptables filter on layer-3 while SIP is on layer-7. It is easily possible > to generate a SIP packet with different source-ip than physical interface. > > > ** ** > > You can also simulate it if you set external-ip=some-else-ip in SIP.com in > asterisk. All you SIP packets will contain new some-else-ip while layer-3 > headers will still have actual physical interface IP. > > > I am usingOS (Elastix distribution). I am not really a champ at system administration hence this went over the top. I will observe the system tonight and send my feedback tomorrow. Thx to everyone for being with me on this. Sans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Firewall Issue
Ok, run your script and then do this: service iptables save And by the way, list "chkconfig --list iptables" output. 2011/8/8 RSCL Mumbai > > On Mon, Aug 8, 2011 at 6:03 PM, Faisal Hanif wrote: > >> If you take a bit deep analyses on SIP packet you will be able to >> understand the issue, >> >> ** ** >> >> Iptables filter on layer-3 while SIP is on layer-7. It is easily possible >> to generate a SIP packet with different source-ip than physical interface. >> >> >> ** ** >> >> You can also simulate it if you set external-ip=some-else-ip in SIP.com in >> asterisk. All you SIP packets will contain new some-else-ip while layer-3 >> headers will still have actual physical interface IP. >> >> >> > I am usingOS (Elastix distribution). > I am not really a champ at system administration hence this went over > the top. > > I will observe the system tonight and send my feedback tomorrow. > > Thx to everyone for being with me on this. > Sans > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CEL and MySQL
cel_odbc.conf and then use adapative odbc I think. -- Thanks, Phil - Original Message - > Is anyone using CEL with a MySQL backed at all? > > I've found a table schema but I'm guessing I need some sort of > cel_mysql.conf and don't even have a sample for that. > > Can anyone give me any pointers as to what files I need to change to > get > this logging to my MySQL table? > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CEL and MySQL
I ended up using the patch here https://issues.asterisk.org/jira/browse/ASTERISK-17638?focusedCommentId=180838#comment-180838 and recompiling. It seems to work fine so far On Mon, 2011-08-08 at 17:17 +0100, --[ UxBoD ]-- wrote: > cel_odbc.conf and then use adapative odbc I think. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail to correctly build 1.8.5 ??
On 08/08/2011 07:51 AM, Norbert Zawodsky wrote: Am 08.08.2011 13:38, schrieb Soeren Malchow (MCon): -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norbert Zawodsky Sent: Sunday, August 07, 2011 8:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] fail to correctly build 1.8.5 ?? Hello everybody, I've been using asterisk 1.2 for quite a long time now, but I thought it's time to try a newer version of asterisk. So I downloaded 1.8.5, extracted the tar, ran configure, make, make install ... Everything looks fine (no obvious compile/link errors). But as soon as I start asterisk, it dies with a segfault. I executed asterisk within strace and last action before the segfault was reading cdr_mysql.conf in. I deleted cdr_mysql.conf and asterisk starts normally. I copied cdr_mysql.conf.sample to /etc/asterisk and asterisk starts normally Then I just uncommented the "hostname=" line and asterisk segfaults during startup. So what could be wrong here? I have packages "libmysqlclient-devel" and "libmysqlclient16" installed, both version 5.1.46-2.18 OS is a SuSE 11.3, 64 bit Any hints? Norbert -- Hi Do you see the loaded modules when not using the conf file, somethinglike this ? cdr_mysql.so MySQL CDR Backend 0 res_config_mysql.so MySQL RealTime Configuration Driver 0 app_mysql.so Simple Mysql Interface 0 soeren Hi Soeren, Yes, "module show" shows all three of them. Please check to see if there is an issue open for this problem on https://issues.asterisk.org/jira. If there is not, please open one; an incorrectly formatted configuration file should not result in a segfault. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRV question
Hello, All, I have a question about using SRV record. One of SIP provider is using DNS SRV record. If I use IP address of the SIP proxy server I can successfully register my Asterisk 1.8.5. But If I try to use the domain name like: /register => user:p...@somedomain.com/ then the registration failed. So I have to "/dig somedomain.com SRV/" to find out the SIP proxy host, for example, "sip.regserver.com", then I have to run "/dig sip.regserver.com/" to find out the real IP address. Does Asterisk has the feature so I can register by SRV record? I lookup up sip.conf and I enabled srvlookup=yes, but that is only for SIP URI outgoing call, not for SIP registration. Thank you in advance. Jian -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more CDR record for simple Hangup?
On 11-08-06 10:06 AM, Miguel Molina wrote:
El 05/08/11 13:20, J Gao escribió:
I am using the new 1.8.5 and I just found out that Asterisk won't
record the call if the call just hangup. I did a test like this:
exten => 1009, 1, Hangup()
Then I called 1009:
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [1009@init-1005:1] Hangup("SIP/1007-003c", "")
in new stack
== Spawn extension (init-1005, 1009, 1) exited non-zero on
'SIP/1005-003c'
I am not sure why now Asterisk doesn't write this into CDR. In the
previous version Asterisk record the hangup call.
Is there anyway I can have the hangup write into the CDR?
What is your cdr.conf configuration?
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here is my cdr.conf:
==
[general]
endbeforehexten=yes
[csv]
usegmtime=yes; log date/time in GMT. Default is "no"
loguniqueid=yes ; log uniqueid. Default is "no"
loguserfield=yes ; log user field. Default is "no"
accountlogs=yes ; create separate log file for each account code.
Default is "yes"
--
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Re: [asterisk-users] No more CDR record for simple Hangup?
El 08/08/11 11:46, J Gao escribió:
On 11-08-06 10:06 AM, Miguel Molina wrote:
El 05/08/11 13:20, J Gao escribió:
I am using the new 1.8.5 and I just found out that Asterisk won't
record the call if the call just hangup. I did a test like this:
exten => 1009, 1, Hangup()
Then I called 1009:
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [1009@init-1005:1] Hangup("SIP/1007-003c", "")
in new stack
== Spawn extension (init-1005, 1009, 1) exited non-zero on
'SIP/1005-003c'
I am not sure why now Asterisk doesn't write this into CDR. In the
previous version Asterisk record the hangup call.
Is there anyway I can have the hangup write into the CDR?
What is your cdr.conf configuration?
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here is my cdr.conf:
==
[general]
endbeforehexten=yes
[csv]
usegmtime=yes; log date/time in GMT. Default is "no"
loguniqueid=yes ; log uniqueid. Default is "no"
loguserfield=yes ; log user field. Default is "no"
accountlogs=yes ; create separate log file for each account code.
Default is "yes"
Try this cdr.conf general setting: unanswered=yes
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Re: [asterisk-users] Polycom and auto answer
Here's what I have and it works for me in 1.8.5: in sip.cfg in extensions.conf exten => 3500,1,SIPAddHeader(Alert-Info: Auto Answer) exten => 3500,n,Page(SIP/3011&SIP/3021&SIP/3110) ; Shortened for example. I actually have about 20 phones here. exten => 3500,n,Hangup This is working fine in an environment with many 330s, some 450s, some 335s and a 550 all running 3.3.1 Hope this helps you out. Bob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
Hi,
This is still broken in 1.6.2.20. Please see below.
On Fri, May 06, 2011 at 04:27:42PM +, Watkins, Bradley wrote:
> >From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> >boun...@lists.digium.com] On Behalf Of Mark G Thomas
> >Sent: Friday, May 06, 2011 12:02 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [asterisk-users] ael context ~~s~~ in macros broke Dial() U()
> >option in 1.6.2.17.2 and newer
> >
> >Hi,
> >
> >On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote:
> >> On 11-05-05 05:14 PM, Mark G Thomas wrote:
> >> >Hi,
> >> >
> >> >I think this must be a bug introduced with 1.6.2.17.something.
> >> >
> >> >When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or
> >> >1.6.2.18, my AEL Dial() commands with the "U" options fail with the
> >following error:
> >> >
> >> >[May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-
> >existent
> >> > destination for gosub: (Context:screen, Extension:s, Priority:1)
> >> >
> >> You might want to have a look at:
> >> https://issues.asterisk.org/view.php?id=18910
> >
> >Thanks. This is it.
> >
> >If I'm reading this right, it describes the change which broke things for me,
> >but no solution applicable to my Dial() command U flag, which is invoking my
> >AEL GoSub (Macro). Switching the Dials back to the M flag doesn't fix it
> >either.
> >
> >It sure seems to me this change to AEL has had unexpected consequences in
> >terms of breaking things in dialplans.
> >
>
> I was under the impression that this had been fixed, although perhaps it's
> not yet in a release. Is there a chance you try with the latest 1.6.2 branch
> from SVN?
>
> - Brad
In AEL, Dial() with the U flag is still broken.
Reverting to a pre-1.6.2.17.2 pval.c fixes the problem.
[Aug 8 13:36:01] ERROR[24608]: app_stack.c:402 gosub_exec: Attempt to reach a
non-existent destination for gosub: (Context:screen, Extension:s, Priority:1)
My AEL dialplan segment:
--
context internals {
102 => {
Dial(${MARKCELL},30,tgU(screen));
jump s@home-menu;
};
};
macro screen() {
Wait(0.5);
Read(ACCEPT,followme/options,1,,1,20);
if( "${ACCEPT}" = "1" ) {
Background(connecting);
} else {
Set(GOSUB_RESULT=CONTINUE);
};
return;
};
Here's the dialplan it created under 1.6.2.20, but U(screen)
in Dial() tries to call screen,s which doesn't exist.
---
[ Context 'internals' created by 'pbx_ael' ]
'102' => 1. Dial(${MARKCELL},30,tgU(screen)) [pbx_ael]
2. Goto(home-menu,s,1)[pbx_ael]
[ Context 'screen' created by 'pbx_ael' ]
'~~s~~' =>1. Wait(0.5) [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20) [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_25) [pbx_ael]
8. Return() [pbx_ael]
This works, from an earlier version, before the pval.c change:
[ Context 'screen' created by 'pbx_ael' ]
's' =>1. Wait(0.5) [pbx_ael]
2. Read(ACCEPT,followme/options,1,,1,20) [pbx_ael]
3. GotoIf($[ "${ACCEPT}" = "1" ]?4:6) [pbx_ael]
4. Background(connecting) [pbx_ael]
5. Goto(7)[pbx_ael]
6. Set(GOSUB_RESULT=CONTINUE) [pbx_ael]
7. NoOp(Finish if_screen_25) [pbx_ael]
8. Return() [pbx_ael]
--
Mark G. Thomas (m...@misty.com)
Web: http://mgtinternet.com/
Tel: +1-215-512-0112 US: 877-512-0112
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[asterisk-users] 1.8 issues with Local Bridging
I encourage the developers to check this out http://forums.asterisk.org/viewtopic.php?f=1&t=77692&p=161590#p161590 I am calling from behind a NAT, and there is no way to force Asterisk to stay in the path. If the codec is the same as the outbound leg, it always does "Remote bridging", but of course, creates a 1 way audio. I tried everything in the book directrtpsetup=no directmedia=nonat canreinvite=nonat and directrtpsetup=no directmedia=no canreinvite=no But it just behaves different than in 1.6.2 Any ideas how to make sure that the NAT works? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom and auto answer
Turns out this was (drum roll) a router issue. pfSense didn`t work (old Beta of 2.0). I'll try upgrading, but can anyone help me understand how a router can allow phones to work 100% correctly except for Alert-Info messages (this is a hosted PBX environment, but everything except paging works)? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Monday, August 08, 2011 7:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom and auto answer Warren, Thanks, I ended up doing that but it didn't change a thing. I mean, the originating phone does not drop into a conference obviously, but the ringing still goes on and on for 30 secs (my timeout). Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, August 08, 2011 2:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom and auto answer On Sun, Aug 7, 2011 at 9:32 PM, Mike wrote: Hi, [paging] exten => s,1,Verbose(1,paging) exten => s,n,Set(TIMEOUT(absolute)=30) ;to prevent call from being stuck exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) exten => s,n,Page(SIP/sipphone) Try changing the Page() to a Dial() command and see if that makes a difference. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CEL and MySQL
you can try cel_mysql 2011/8/9 Ishfaq Malik : > I ended up using the patch here > > https://issues.asterisk.org/jira/browse/ASTERISK-17638?focusedCommentId=180838#comment-180838 > > and recompiling. It seems to work fine so far > > On Mon, 2011-08-08 at 17:17 +0100, --[ UxBoD ]-- wrote: >> cel_odbc.conf and then use adapative odbc I think. > > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Callerid and transfer problem
Please help. I would be surprised if no one ever faced this problem
On Mon, Aug 8, 2011 at 3:16 PM, Jim Boykin wrote:
> Hi, We need some help.
>
> We are unable to transfer the incoming call from DAHDI to another
> number. We are able to receive calls and dial out fine, but what we
> really want to achieve is to transfer the call so that PRI will be
> free and also the transferred number will get the received callerid.
> Here is dialplan.
>
> [default]
> exten => 1999001,1,myprint(${CALLERID(num)})
> exten => 1999001,n,transfer(DAHDI/1/14085551234)
> exten => 1999001,n,myprint(${TRANSFERSTATUS})
>
> The problem is transfer fails and TRANSFERSTATUS is set to
> UNSUPPORTED. It works if we change 'transfer' command to 'Dial' but in
> this case it does not pass callerid. facilityenable and transfer is
> set to 'yes' in chan_dahdi.conf.
>
> Any hints what we are doing wrong?
>
> Thanks
> Jim
>
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