[asterisk-users] Odd Local Channel and 0 billsec issue
Hi
I've been noticing an odd issue with our servers (1.4.17) where a large
number of one particular customer's (we operate a hosted VoIP platform)
calls go through a Local channel rather than the SIP channel and
whenever this happens our asterisk CDR is recording a billsec value of 0.
Our outgoing calls to POTS are sent through a separate carrier and we
get a daily CDR off them in which these same calls have a non 0 duration
so we are obviously making a loss on these calls.
All out customers outgoing calls go through the same macro which is as
follows
[macro-extcall]
;Macro created by Ish to handle external calls
exten => s,1,Set(CALLERID(all)=${ARG2})
exten => s,2,Dial(SIP/44${ar...@carrier)
exten => s,3,Hangup
exten => s,102,Playtones(busy)
exten => s,103,Congestion
I've seen the same issue very occasionally with other customers but with
one particular customer a large proportion, but not all the calls show
this issue.
Has anyone had any experience of similar issues?
Thanks in advance
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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[asterisk-users] Kamailio and asterisk Integration
Dear All, I Have one test scenario where one of my kamailio servers is in Europe and Another is in Singapore. extensions can be registered to any of these 2 kamailio servers in the same domain. now i want to send a call to an extension from Asterisk to one of these kamailio servers based on extension registered to it. is there any easy solution to get this done? regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file
Well, how about piping an internet stream into a phone call via some app in an extension? Alex Balashov wrote: > cov...@ccs.covici.com wrote: > > > Is there any app to pipe a stream to a call either a meetme conference > > or even a regular call? > > Do you mean piping outside audio of some description into a MeetMe > conference? > > If so, I do not know if there is a pre-built app, but this can be > achieved relatively trivially with the use of a well-placed AMI > Originate command (or, perhaps, call files) combined with Local dial > plan channels. > > -- Alex > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write the incoming stream to pipe/socket instead of .gsm file
It seems to me this could be achieved in the manner I just described by specifying an Internet stream as a music on hold source and putting the outgoing channel on hold via a Local channel. -- Sent from mobile device On Nov 16, 2009, at 5:36 AM, cov...@ccs.covici.com wrote: > Well, how about piping an internet stream into a phone call via some > app > in an extension? > > Alex Balashov wrote: > >> cov...@ccs.covici.com wrote: >> >>> Is there any app to pipe a stream to a call either a meetme >>> conference >>> or even a regular call? >> >> Do you mean piping outside audio of some description into a MeetMe >> conference? >> >> If so, I do not know if there is a pre-built app, but this can be >> achieved relatively trivially with the use of a well-placed AMI >> Originate command (or, perhaps, call files) combined with Local dial >> plan channels. >> >> -- Alex >> >> -- >> Alex Balashov - Principal >> Evariste Systems >> Web : http://www.evaristesys.com/ >> Tel : (+1) (678) 954-0670 >> Direct : (+1) (678) 954-0671 >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > cov...@ccs.covici.com > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kamailio and asterisk Integration
I suppose that would depend on how the information about the registrations is organised; do you want Asterisk to query some sort of database used for backing these registrars and figure out where the contact binding for a given AOR resides? AGI and func_odbc provide fine ways to do that. As a practical matter, the simplest solution is to send the call to both registrars - if users only register to one at a time. The one that the user is not registered to will return a 404 Not Found and the other will return call progress. The only price to pay is a little extra bandwidth per call for the additional signaling, but you also save unnecessary database queries, script invocations, and/or other process related to the resolution of the appropriate registrar. This does not sound like very good design, however. If I am inferring your intent correctly from your post, you are placing high-performance nodes (Kamailio) at the access edge and using a low-performance element in the central core (Asterisk), relatively speaking. Also, there should be some logic to the use of particular registrars for particular users; it should not simply be an arbitrary choice, and you should not need a "meta-location" server. If there is logic, you can route based on that logic. -- Sent from mobile device On Nov 16, 2009, at 5:16 AM, DHAVAL INDRODIYA wrote: > Dear All, > > I Have one test scenario where one of my kamailio servers is in > Europe and Another is in Singapore. > > extensions can be registered to any of these 2 kamailio servers in > the same domain. now i want to send a call to an extension from > Asterisk to one of these kamailio servers based on extension > registered to it. > > is there any easy solution to get this done? > > regards > > > > > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with dahdi on asterisk 1.6.1.9 with TE122
Hello, I am installing dahdi on a machine lspci 00:00.0 Host bridge: Intel Corporation 3200/3210 Chipset DRAM Controller (rev 01) 00:01.0 PCI bridge: Intel Corporation 3200/3210 Chipset Host-Primary PCI Express Bridge (rev 01) 00:06.0 PCI bridge: Intel Corporation 3210 Chipset Host-Secondary PCI Express Bridge (rev 01) 00:1a.0 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #5 (rev 02) 00:1a.2 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #6 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801I (ICH9 Family) USB2 EHCI Controller #2 (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801I (ICH9 Family) PCI Express Port 1 (rev 02) 00:1c.4 PCI bridge: Intel Corporation 82801I (ICH9 Family) PCI Express Port 5 (rev 02) 00:1c.5 PCI bridge: Intel Corporation 82801I (ICH9 Family) PCI Express Port 6 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801I (ICH9 Family) USB UHCI Controller #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801I (ICH9 Family) USB2 EHCI Controller #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev 92) 00:1f.0 ISA bridge: Intel Corporation 82801IR (ICH9R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801IR/IO/IH (ICH9R/DO/DH) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801I (ICH9 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801I (ICH9 Family) 2 port SATA IDE Controller (rev 02) 00:1f.6 Signal processing controller: Intel Corporation 82801I (ICH9 Family) Thermal Subsystem (rev 02) 01:00.0 PCI bridge: Intel Corporation 6700PXH PCI Express-to-PCI Bridge A (rev 09) 01:00.1 PIC: Intel Corporation 6700/6702PXH I/OxAPIC Interrupt Controller A (rev 09) 01:00.2 PCI bridge: Intel Corporation 6700PXH PCI Express-to-PCI Bridge B (rev 09) 01:00.3 PIC: Intel Corporation 6700PXH I/OxAPIC Interrupt Controller B (rev 09) 03:02.0 Ethernet controller: Digium, Inc. Wildcard TE122 single-span T1/E1/J1 card (rev 11) 04:00.0 RAID bus controller: 3ware Inc 9650SE SATA-II RAID (rev 01) 05:00.0 PCI bridge: Texas Instruments XIO2000(A)/XIO2200(A) PCI Express-to-PCI Bridge (rev 03) 06:08.0 Ethernet controller: Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) (rev 11) 0d:00.0 Ethernet controller: Intel Corporation 82573E Gigabit Ethernet Controller (Copper) (rev 03) 0f:00.0 Ethernet controller: Intel Corporation 82573L Gigabit Ethernet Controller 11:04.0 VGA compatible controller: ATI Technologies Inc ES1000 (rev 02) dahidi 2.2.0.2 compiles fine. I am running Gentoo x64. The Service starts fine. lsmod |grep dahdi dahdi_echocan_mg2 4864 0 dahdi_vpmadt032_loader 176560 0 dahdi_voicebus 28928 2 dahdi_vpmadt032_loader,wcte12xp dahdi 185616 3 dahdi_echocan_mg2,wcte12xp,dahdi_voicebus Wem starting asterisk 1.6.1.9 I get the message: ...Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection. You have options: 1. You only have to compile DAHDI support into Asterisk if you need it. One option is to recompile without DAHDI support. 2. You only have to load DAHDI drivers if you want to take advantage of DAHDI services. One option is to unload DAHDI modules if you don't need them. cat /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Mon Nov 16 11:46:50 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Global data loadzone= de defaultzone = de cat /etc/asterisk/chan_dahdi.conf [channels] language=de group=1 signalling=pri_net switchtype=euroisdn context=default ;rxgain=-4.0 ;txgain=-4.0 channel => 1-15,17-31 pridialplan=unknown echocancel=yes What can I do? Thanks, Oliver -- Schröder Assistance und Consulting GmbH Lohdiecksweg 6 59457 Werl Fon 02922-8037-490 Fax 02922-8037-540 -- Sitz der Gesellschaft: Werl HRB 6747 Amtsgericht Arnsberg Geschäftsführer: Lutz Schröder Steuernummer: 5343/5708/1703 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Security Against brute force attack
Has Asterisk any protection against brute force attack for SIP authentication? Something like a maximum login attempt limit Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.18-rc3: SendFAX causes restart
sean darcy wrote:
> Leif Madsen wrote:
>> sean darcy wrote:
>>> On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
>>> asterisk restarts:
>>>
>>> Before I file a bug, is there anything I'm missing?
>> Does this happen on earlier versions of the 1.6.0 series prior to this
>> release
>> candidate? I'm curious if this is a regression, or if this is an existing
>> bug.
>>
>> Leif.
>>
>
> Don't know - just trying faxing now.
>
> In any event, I rebuilt 1.6.0.18-rc3, and now it doesn't crash, but it
> still doesn't work:
>
> -- Executing [...@fax-tx-test:3]
> SendFAX("SIP/nhi-riverside-sip-",
> "/var/spool/asterisk/fax/20091113_1455.tif") in new stack
> [Nov 15 19:28:59] WARNING[4520]: app_fax.c:178 phase_e_handler: Error
> transmitting fax. result=49: The call dropped prematurely.
> [Nov 15 19:28:59] WARNING[4520]: app_fax.c:762 transmit: Transmission failed
> -- Executing [...@fax-tx-test:4]
> Hangup("SIP/nhi-riverside-sip-", "") in new stack
>== Spawn extension (fax-tx-test, s, 4) exited non-zero on
> 'SIP/nhi-riverside-sip-'
>
> On the receive side:
>
> -- Executing [...@incoming-fax:2]
> ReceiveFAX("SIP/nhi-riverside-sip-0017",
> "/var/spool/asterisk/fax/20091115_1928.tif") in new stack
> [2009-11-15 19:28:59] WARNING[3923]: app_fax.c:178 phase_e_handler:
> Error transmitting fax. result=48: Disconnected after permitted retries.
> [2009-11-15 19:28:59] ERROR[3923]: app_fax.c:700 transmit_t38: channel
> 'SIP/nhi-riverside-sip-0017' failed to disable T.38
> [2009-11-15 19:28:59] WARNING[3923]: app_fax.c:762 transmit:
> Transmission failed
> -- Executing [...@incoming-fax:3]
> Hangup("SIP/nhi-riverside-sip-0017", "") in new stack
>== Spawn extension (incoming-fax, s, 3) exited non-zero on
> 'SIP/nhi-riverside-sip-0017'
>
> The receive fax side works fine receiving faxes generally.
>
> sean
>
Well I spoke too soon, it does crash. Tried it over PSTN instead of SIP:
[Nov 15 21:48:10] VERBOSE[5701] logger.c: -- Executing
[...@fax-tx-test:3] ESC[1;36;40mSendFAXESC[0;37;40m("ESC[1;35;40mDAHDI/1-1
ESC[0;37;40m",
"ESC[1;35;40m/var/spool/asterisk/fax/20091113_1455.tifESC[0;37;40m") in
new stack
[Nov 15 21:48:37] VERBOSE[5701] logger.c: -- Executing
[...@fax-tx-test:4] ESC[1;36;40mHangupESC[0;37;40m("ESC[1;35;40mDAHDI/1-1
ESC[0;37;40m", "ESC[1;35;40mESC[0;37;40m") in new stack
[Nov 15 21:48:41] VERBOSE[5723] logger.c: Asterisk Event Logger Started
/var/log/asterisk/event_log
Is anyone else seeing this?
sean
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Re: [asterisk-users] Database postgresql not able to start
I found that on a clean boot, I could not connect to Postgresql either. In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the module, and that seems to work. After bootup, cdr_pgsql.so is able to connect immediately. -- James Texter III Sr. Software Engineer NOBLE SYSTEMS 4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452 (o) 404.851.1331 ext. 357 (f) 404.851.1421 (e) jtext...@noblesys.com (w) www.noblesys.com We succeed when we exceed our customers expectations! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, November 15, 2009 4:09 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Database postgresql not able to start On Sun, Nov 15, 2009 at 06:52:22AM +, aster...@opensourcesolution.in wrote: > > > i have installed database POSTGRESQL for storing call details. when i > restart database i get the error. > > [r...@localhost server]# psql -h > 127.0.0.1 -U asterisk Password > psql: could not connect to server: > Connection refused > Is the server running on host "127.0.0.1" and > accepting > TCP/IP connections on port 5432? I'm not sure about other systems. On my systems working as root is not the simple way to work with database. Try 'su - postgres' and connect as that user. It should work as a PostgreSQL admin user. > > THIS IS MY > /VAR/LIB/PGSQL/DATA/POSTGRESQL.CONF > > # CONNECTIONS AND > AUTHENTICATION Small hint: Text in ALL CAPS is generally considered as shouting. Please try to avoid that if you don't really need it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CONFIDENTIAL NOTICE : If you have received this email in error, please immediately notify the sender by email at the address shown above. This email may contain confidential or legally privileged information that is intended only for the use of the individual or entity named in this email. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or reliance upon the contents of this email is strictly prohibited. Please delete from your files if you are not the intended recipient. Thank you for your compliance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Database postgresql not able to start
James Texter wrote: > I found that on a clean boot, I could not connect to Postgresql either. > In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the > module, and that seems to work. After bootup, cdr_pgsql.so is able to > connect immediately. > This sounds as though you have Asterisk starting before PostgreSQL. If you're using CentOS or RHEL or other RHEL-inspired distro look at /etc/init.d/asterisk and /etc/init.d/postgresql. Compare the line in each that looks like: # chkconfig: 2345 xx yy The 'xx' is the start priority. If the number is lower in the asterisk file than it is in the postgresql file then that's your problem. You need PG to start before Asterisk. man chkconfig for further details on what you can do. Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ENUM and Asterisk 1.6
Hi all,
I have a problem with 1.6.1.7-rc1 and ENUM (with an own PowerDNS server
and NAPTR record). Maybe somebody has more experience with this or can
give me some input.
The dialplan:
exten => 292,1,Set(DIAL_NUMBER=43660123456)
exten => 292,2,Set(sip=
${ENUMLOOKUP(+${DIAL_NUMBER},sip,,1,ns3.e164.xxx.com)}) ;x'ed out the
domain name starting from here
exten => 292,3,NoOp(${sip})
exten => 292,4,Hangup()
The output if I dial 292:
Connected to Asterisk 1.6.1.7-rc1 currently running on srv21 (pid =
6061)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [...@sip:1] Set("SIP/273-98117048",
"DIAL_NUMBER=43660123456") in new stack
== ast_get_enum(num='+43660123456', tech='sip',
suffix='ns3.e164.xxx.com', options='', record=1
== ENUM options(): pos=1, options='0'
== ast_get_enum() profiling: FAIL,
6.5.4.3.2.1.0.6.6.3.4.ns3.e164.xxx.com, 2 ms
-- Executing [...@sip:2] Set("SIP/273-98117048", "sip=") in new
stack
-- Executing [...@sip:3] NoOp("SIP/273-98117048", "") in new stack
-- Executing [...@sip:4] Hangup("SIP/273-98117048", "") in new stack
== Spawn extension (sip, 292, 4) exited non-zero on 'SIP/273-98117048'
-- Executing [...@sip:1] Hangup("SIP/273-98117048", "") in new stack
== Spawn extension (sip, h, 1) exited non-zero on 'SIP/273-98117048'
srv21*CLI>
The NAPTR record:
6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com 100 10 "u" "E2U+sip" "!^.*$!
sip:2...@10.0.43.105!" .
The output of a dig command using the ns3.e164.xxx.com (so DNS seems to
be fine):
dig @ns3.e164.xxx.com 6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com ANY
; <<>> DiG 9.5.1-P3 <<>> @ns3.e164.xxx.com
6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com ANY
; (1 server found)
;; global options: printcmd
;; Got answer:
;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 11934
;; flags: qr aa rd; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 0
;; WARNING: recursion requested but not available
;; QUESTION SECTION:
;6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com. IN ANY
;; ANSWER SECTION:
6.5.4.3.2.1.0.6.6.3.4.e164.xxx.com. 600 IN NAPTR 100 10 "u" "E2U+sip" "!
^.*$!sip:2...@10.0.43.105!" .
;; Query time: 19 msec
;; SERVER: 10.0.50.107#53(10.0.50.107)
;; WHEN: Mon Nov 16 14:14:05 2009
;; MSG SIZE rcvd: 111
enum.conf:
[general]
;
; The search list for domains may be customized. Domains are searched
; in the order they are listed here.
;
search => ns3.e164.xxx.com
search => e164.arpa
Regards,
Erik
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Re: [asterisk-users] Database postgresql not able to start
I thought that too, but I already checked. Postgresql is priority 64, Asterisk is priority 90. Watching the boot sequence, I can see that Postgresql is clearly started before Asterisk. It may be that there is something in my config that causes this happen, but it's reproducable on any box I setup. -- James Texter III Sr. Software Engineer NOBLE SYSTEMS 4151 Ashford Dunwoody Road, Suite 600 | Atlanta, GA 30319-1452 (o) 404.851.1331 ext. 357 (f) 404.851.1421 (e) jtext...@noblesys.com (w) www.noblesys.com We succeed when we exceed our customers expectations! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline Sent: Monday, November 16, 2009 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Database postgresql not able to start James Texter wrote: > I found that on a clean boot, I could not connect to Postgresql either. > In my /etc/rc.local, I unload cdr_pgsql.so, sleep 15, then reload the > module, and that seems to work. After bootup, cdr_pgsql.so is able to > connect immediately. > This sounds as though you have Asterisk starting before PostgreSQL. If you're using CentOS or RHEL or other RHEL-inspired distro look at /etc/init.d/asterisk and /etc/init.d/postgresql. Compare the line in each that looks like: # chkconfig: 2345 xx yy The 'xx' is the start priority. If the number is lower in the asterisk file than it is in the postgresql file then that's your problem. You need PG to start before Asterisk. man chkconfig for further details on what you can do. Barry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CONFIDENTIAL NOTICE : If you have received this email in error, please immediately notify the sender by email at the address shown above. This email may contain confidential or legally privileged information that is intended only for the use of the individual or entity named in this email. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or reliance upon the contents of this email is strictly prohibited. Please delete from your files if you are not the intended recipient. Thank you for your compliance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-Site GUI
Most of the OS solutions I've seen require PHP and MYSQL. If you set up a daemon/script to feed to mysql on a common server, most of this would work fine. From my experience, I'd prefer some sort of central script (C/Perl/PHP) that uses AMI to talk to the other boxes. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Saturday, November 14, 2009 3:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Multi-Site GUI I am searching for a GUI platform that can support multiple sites with Asterisk servers with a single GUI to manage them all. I have something currently working but it is not as pretty or polished as I would like, although it does work quite well. Are there any opensource options out there? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip source aware Authentication
Alex Balashov wrote: > As far as I know, Asterisk has no way to restrict the content of the > domain portion of the Contact URI. However, most commercial SBCs > should have a way to filter this, and it is highly recommended that > you do so. It does actually; we added it to address this very issue. This also keeps people from registering Contact URIs using RFC-1918 addresses that happen to live inside the ITSP's network, or other nastiness. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor and Call Latency during conversation
Hi,
We are using MixMonitor to record the call. When the call is bridged, the
latency is significant. We tried to increase the internet speed and the
server RAM and processor speed and still we are having that issue.
We use VoiceTrading and Gafachi's Termination minutes to make calls. As we
are in US and VoiceTrading in Europe, somebody suggested to move the
termination minute provider to within USA. So, we bought the minutes from
Gafachi. Still we are having the call latency issues.
$ConversationFile =
$ConversationPath."conv_"."$CallQID-$ConversationID.wav";
$self->agi->answer();
$self->agi->exec("MixMonitor", "$ConversationFile|ba");
Any suggestions would be greatly appreciated.
Thanks a lot
Bharath
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[asterisk-users] asterisk cdr - remote ip address
hi, i want add info about remote party ip address to the asterisk cdr table can you recommend me "the system way"? thanks --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk cdr - remote ip address
- exten => s,x,Set(CDR(userfield) = "information") - replace "information"
with the information like ${remoteip}
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
Sent: Monday, November 16, 2009 8:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk cdr - remote ip address
hi,
i want add info about remote party ip address to the asterisk cdr table
can you recommend me "the system way"?
thanks
---
Marek Cervenka
===
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Re: [asterisk-users] MixMonitor and Call Latency during conversation
On Mon, Nov 16, 2009 at 9:40 AM, Bharath B. Reddy Bynagari
wrote:
> We are using MixMonitor to record the call. When the call is bridged, the
> latency is significant.
> $ConversationFile =
> $ConversationPath."conv_"."$CallQID-$ConversationID.wav";
>
> $self->agi->answer();
>
> $self->agi->exec("MixMonitor", "$ConversationFile|ba");
You're obviously using SIP. I don't like to admit it, but I've seen
this problem before.
Please try modifying the voice-activity-detection sections of your SIP
settings and see if this fixes the problems.
My hunch, which is not proven, is that when SIP silence detection
thinks it should stop transmitting packets, the recording module
thinks it shouldn't record the lack of voice transmission, and then
the timing in the recording gets farther and farther from the truth
the longer the call goes on.
in asterisk.conf
transmit_silence = yes
transmit_silence_during_record = yes
in dsp.conf
silencethreshold=1000
in codecs.conf
vad => false
pp_vad => false
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Re: [asterisk-users] Changing labels on Phones
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian > > We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a > "hotdesk" type system where anyone can log on to an extension - however what > I would love to do is relabel the phone with the current "owner" when this > logon happens. I know that I can change the sip.conf and phones tftp file, > however this is a big problem with the Cisco's as they take *forever* (ok, > maybe 2 / 3 minutes) to reboot (VLAN problem) > 1) Has anyone actually solved this VLAN issue with the cisco ? > 2) Is there any way of changing a label without rebooting the phone ? > TIA I have not personally tried this, but I remember someone had posted a way to script the change of the background image on Cisco 79x1 phones. You could create a dynamic image in PHP that had the user info on it, then kick off the script to change the background image. Might be a little tricky, but no reboot required! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing labels on Phones
I'm pretty sure it only pulls the background image during a reboot. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: Monday, November 16, 2009 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Changing labels on Phones > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian > > We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a > "hotdesk" type system where anyone can log on to an extension - however what > I would love to do is relabel the phone with the current "owner" when this > logon happens. I know that I can change the sip.conf and phones tftp file, > however this is a big problem with the Cisco's as they take *forever* (ok, > maybe 2 / 3 minutes) to reboot (VLAN problem) > 1) Has anyone actually solved this VLAN issue with the cisco ? > 2) Is there any way of changing a label without rebooting the phone ? > TIA I have not personally tried this, but I remember someone had posted a way to script the change of the background image on Cisco 79x1 phones. You could create a dynamic image in PHP that had the user info on it, then kick off the script to change the background image. Might be a little tricky, but no reboot required! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121 - Idle system load at ~0.3 - Bad DAHDI 2.2.0.2 behaviour ?!
Shaun, Thanks for your feedback. See my inline comments. On Fri, Nov 13, 2009 at 7:18 PM, Shaun Ruffell wrote: > > It appears there may be a regression in dahdi-linux 2.2.0 with regards to > the wcte12xp driver and the VPMADT032 module (as discussed > https://issues.asterisk.org/view.php?id=15724). Would you be willing to try > at least revision 7584 of > http://svn.asterisk.org/svn/dahdi/linux/branches/2.2 and report your results > on that issue? > We will, either on 15724 or on 15798, both, I'd say, closely related to what we're experiencing. First, however, we will test 2.1.0.4 for at least 10 days to effectively confirm are experiencing the regression you refer. We defined 10 stable days because we've been experiencing 1 - 3 failures per week with 2.2.0.2. So, we will move to 2.1.0.4 later today and, if there are no incidents, we will move to SVN revision 7584 or later on Nov 27th. > > The idle load you're seeing can be a little misleading, but essentially, > once you load the drivers for both the wctdm24xxp and wcte12xp, there is a > fixed cost associated with continuously moving the TDM data to/from the > card. The load imposed by the drivers would only go up after this point if > a) software echocan is enabled, or b) you're conferencing many calls in the > kernel. Otherwiseit's fixed. > I agree the load can be misleading however, consider: - The system is really idle - zero calls, zero activity, no other software. - 0.3 to 0.4 load on a modern CPU is a lot of processing, is there that much data to move around ?! :) We tested some variations and here is what we found: (recall, AEX410 with no VPM is physically installed) DAHDI 2.2.0.2 - Removed TE121 - Idle load is 0 - TE121 without VPM - Idle load is 0.3 - 0.4 - TE121 with VPM - Idle load is 0.3 - 0.4 DAHDI 2.1.0.4 - TE121 without VPM - Idle load is 0.01 - 0.02 - TE121 with VPM - Idle load is 0.01 - 0.05 DAHDI 2.2.0.1 - TE121 with VPM - Idle load is 0.26 - 0.32 DAHDI 2.2.0 - TE121 with VPM - Idle load is 0.31 - 0.36 DAHDI SVN-branch-2.2-r7584 - TE121 with VPM - Idle load is 0.69 - 0.82 Under all cases, we stopped asterisk, unloaded DAHDI, rebuilt DAHDI (+Asterisk if needed), loaded DAHDI, started Asterisk, waited 120s, tested one inbound call + one outbound call. In short: - DAHDI 2.1, Idle load is 0 - DAHDI 2.2, Idle load is 0.3 - 0.4 - DAHDI 2.2. SVN, Idle load is roughly twice as in 2.2 releases So, while your explanation regarding system load makes sense, I find it odd that 2.2, which apparently is not working ok for us in this case, is showing such behaviour when 2.1 does not and neither Zaptel 1.2/1.4 used many other systems I've installed. What would you say the explanation for these observations is ? Would you say that there is a correlation between the load and the suspected not so good behaviour we're getting from DAHDI 2.2.0.2 on our case ? Or is it just a coincidence ? Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing labels on Phones
On Mon, Nov 16, 2009 at 7:29 AM, Peder wrote: > I'm pretty sure it only pulls the background image during a reboot. On a 79x0, yes. On the 79x1 phones the user can change the background to a list of custom images that you provide. It downloads the image on the fly, and applies it. Here is a link to a reboot script http://www.dave.vc/wordpress/?p=14 that uses your ability to "press keys" on the phone. You could apply the same idea to "press" the correct buttons to change the background without rebooting. I can't find the script that I found to do this, but I'll keep looking when I get a chance. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing labels on Phones
If you just want to change the SIP configuration on the phone remotely there are a few prerequisites: 1) Upgrade your phone's firmware to a recent release. In the phone's config file on the tftp server: 2) Set telnet_level set to 2. Make sure it stays at 2 when you create your new config file for the phone, otherwise you won't be able to redo this via telnet. 3) You need a phone_password set. If the phone was not loaded with the telnet_level at 2, you will need to reboot the phone manually first for this method to work. With those two items out of the way, you (or your script/program) can telnet to the phone and issue an "erase protflash" which will cause the phone to erase the current SIP configuration and re-read the SIP config files only - without the painful reboot. Takes about 11 seconds on my old 7960 to do. There are some security risk associated with the above, but if you understand them and live with them or can mitigate them in other ways, you're set. -Elliot -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: Monday, November 16, 2009 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Changing labels on Phones > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian > > We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a > "hotdesk" type system where anyone can log on to an extension - however what > I would love to do is relabel the phone with the current "owner" when this > logon happens. I know that I can change the sip.conf and phones tftp file, > however this is a big problem with the Cisco's as they take *forever* (ok, > maybe 2 / 3 minutes) to reboot (VLAN problem) > 1) Has anyone actually solved this VLAN issue with the cisco ? > 2) Is there any way of changing a label without rebooting the phone ? > TIA I have not personally tried this, but I remember someone had posted a way to script the change of the background image on Cisco 79x1 phones. You could create a dynamic image in PHP that had the user info on it, then kick off the script to change the background image. Might be a little tricky, but no reboot required! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is intended only for the use of the individual (s) or entity to which it is addressed and may contain information that is privileged, confidential, and/or proprietary to Calling Circles LLC and its affiliates. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution, forwarding or copying of this communication is prohibited without the express permission of the sender. If you have received this communication in error, please notify the sender immediately and delete the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk cdr - remote ip address
> - exten => s,x,Set(CDR(userfield) = "information") - replace "information"
> with the information like ${remoteip}
${remoteip} variable doesnt exist in asterisk (for remote voip phone)
SIPURI=sip:6...@192.168.1.184:5061 doesnt have public ip
i'm only found way
- check ${CHANNEL} for name
- check astDB SIP/Registry
- set some variable
really doesnt exist some "cleaner" way?
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenka
> Sent: Monday, November 16, 2009 8:50 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] asterisk cdr - remote ip address
>
> hi,
>
> i want add info about remote party ip address to the asterisk cdr table
>
> can you recommend me "the system way"?
---
Marek Cervenka
===
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Re: [asterisk-users] Changing labels on Phones
Here is a link to a reboot script http://www.dave.vc/wordpress/?p=14 that uses your ability to "press keys" on the phone. You could apply the same idea to "press" the correct buttons to change the background without rebooting. I can't find the script that I found to do this, but I'll keep looking when I get a chance. -Jonathan Here is the snippet of code that I use to set the image on the 79x1 series (should plug right into my reboot script via the link posted by Jonathan): $actions['setimage'][0] = array(0 => "Key:Settings", 1 => ".3"); $actions['setimage'][1] = array(0 => "Key:KeyPad1", 1 => ".3"); $actions['setimage'][2] = array(0 => "Key:KeyPad2", 1 => "5"); $actions['setimage'][3] = array(0 => "Key:KeyPad2", 1 => "3"); $actions['setimage'][4] = array(0 => "Key:Soft1", 1 => "7"); $actions['setimage'][5] = array(0 => "Key:Soft2", 1 => "1"); $actions['setimage'][6] = array(0 => "Key:Soft3", 1 => "1"); $actions['setimage'][7] = array(0 => "Key:Soft3", 1 => "1"); And here is how I set the ringer: $actions['regularring'][0] = array(0 => "Key:Settings", 1=> ".3"); $actions['regularring'][1] = array(0 => "Key:KeyPad1", 1 => ".3"); $actions['regularring'][2] = array(0 => "Key:KeyPad1", 1 => ".3"); $actions['regularring'][3] = array(0 => "Key:KeyPad1", 1 => "4"); $actions['regularring'][4] = array(0 => "Key:KeyPad1", 1 => ".1"); $actions['regularring'][6] = array(0 => "Key:Soft1", 1 => "1"); $actions['regularring'][7] = array(0 => "Key:Soft3", 1 => "1"); $actions['regularring'][8] = array(0 => "Key:Settings", 1 => ".1"); I guess I should update my blog with the new version :) *note*: These key sequences are correct on 8-4-1S, I know they have changed in previous firmware changes and chances are they have/will change with versions beyond 8-4-1S. Cheers Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Local Channel and 0 billsec issue
CLI output of calls that go through the local channel instead of the defined
channel would be useful to help diagnose what's going on here.
Thanks,
--Warren Selby
On Mon, Nov 16, 2009 at 4:01 AM, Ishfaq Malik wrote:
> Hi
>
> I've been noticing an odd issue with our servers (1.4.17) where a large
> number of one particular customer's (we operate a hosted VoIP platform)
> calls go through a Local channel rather than the SIP channel and
> whenever this happens our asterisk CDR is recording a billsec value of 0.
>
> Our outgoing calls to POTS are sent through a separate carrier and we
> get a daily CDR off them in which these same calls have a non 0 duration
> so we are obviously making a loss on these calls.
>
> All out customers outgoing calls go through the same macro which is as
> follows
> [macro-extcall]
> ;Macro created by Ish to handle external calls
> exten => s,1,Set(CALLERID(all)=${ARG2})
> exten => s,2,Dial(SIP/44${ar...@carrier)
> exten => s,3,Hangup
> exten => s,102,Playtones(busy)
> exten => s,103,Congestion
>
> I've seen the same issue very occasionally with other customers but with
> one particular customer a large proportion, but not all the calls show
> this issue.
>
> Has anyone had any experience of similar issues?
>
> Thanks in advance
>
> Ish
> --
> Ishfaq Malik
> Software Developer
> PackNet Ltd
>
> Office: 0161 660 3062
>
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Thanks,
--Warren Selby
http://www.selbytech.com
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[asterisk-users] ODP: Re: Changing labels on Phones
It is not the VLAN problem. Simply reboot of the 79xx takes up to 3 minutes and we found no way to speed it up (phones works with Call Manager and VLAN's are not implemented). There are some other methods to display content on the phone screen without editing local configs. Check http://www.ciptec.co.uk/ - commercial site but shows the way. Jacek - Wiadomość oryginalna - Od:: Elliot Otchet Data:: Poniedziałek, 16 Listopad 2009 17:35 Temat: Re: [asterisk-users] Changing labels on Phones > If you just want to change the SIP configuration on the phone > remotely there are a few prerequisites: > > 1) Upgrade your phone's firmware to a recent release. > In the phone's config file on the tftp server: > 2) Set telnet_level set to 2. Make sure it stays at 2 when you > create your new config file for the phone, otherwise you won't be > able to redo this via telnet. > 3) You need a phone_password set. > If the phone was not loaded with the telnet_level at 2, you will > need to reboot the phone manually first for this method to work. > > With those two items out of the way, you (or your script/program) > can telnet to the phone and issue an "erase protflash" which will > cause the phone to erase the current SIP configuration and re-read > the SIP config files only - without the painful reboot. Takes > about 11 seconds on my old 7960 to do. > > There are some security risk associated with the above, but if you > understand them and live with them or can mitigate them in other > ways, you're set. > > -Elliot > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk- > users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman > Sent: Monday, November 16, 2009 10:20 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Changing labels on Phones > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian > > > > We have several types of phones, Cisco 79xx, Aastra 9133i etc. > We have a > > "hotdesk" type system where anyone can log on to an extension - > however what > > I would love to do is relabel the phone with the current "owner" > when this > > logon happens. I know that I can change the sip.conf and phones > tftp file, > > however this is a big problem with the Cisco's as they take > *forever* (ok, > > maybe 2 / 3 minutes) to reboot (VLAN problem) > > 1) Has anyone actually solved this VLAN issue with the cisco ? > > 2) Is there any way of changing a label without rebooting the > phone ? > > TIA > > I have not personally tried this, but I remember someone had > posted a > way to script the change of the background image on Cisco 79x1 phones. > You could create a dynamic image in PHP that had the user info on it, > then kick off the script to change the background image. Might be a > little tricky, but no reboot required! > > -Jonathan > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com > -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinf > > This message is intended only for the use of the individual (s) or > entity to which it is addressed and may contain information that > is privileged, confidential, and/or proprietary to Calling Circles > LLC and its affiliates. If the reader of this message is not the > intended recipient, you are hereby notified that any > dissemination, distribution, forwarding or copying of this > communication is prohibited without the express permission of the > sender. If you have received this communication in error, please > notify the sender immediately and delete the original message. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com > -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinf > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Security Against brute force attack
fail2ban http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk 2009/11/16 Xavier Mesquida > Has Asterisk any protection against brute force attack for SIP > authentication? > Something like a maximum login attempt limit > Thanks > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODP: Re: Changing labels on Phones
There are some other methods to display content on the phone screen without editing local configs. Check http://www.ciptec.co.uk/ - commercial site but shows the way. If you just want to display user info on the phone, why not use the idle url feature: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2007/09/weather_report_for_idle_screen_on_cisco_ip_phones.html In conjunction with a 15 or 30 second meta refresh tag, this seems like it would be able to pull up-to-date info for the phone display periodically. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODP: Re: Changing labels on Phones
You're right - this doesn't sound like VLAN problem (but maybe a VLAN issue is slowing the firmware download down, I can't tell. Can you?). This is a problem of trying to reproduce "extension mobility" like features in CallManager and the like. You're right - rebooting does take a while. In 2001 it was slow. It is still slow today. Yes - you can fake the user out (maybe) by updating images on the phone. Yes you can push content to the phone, or push content that triggers a pull. In 2002, I demonstrated something similar to Ciptec's product at Cisco's IP Telephony Apps Showcase in 2002. (I think I'm still on a DVD interview of that somewhere. Yikes!!) As others have stated, you can use fun features like idle_url, the services button, or push mechanisms to change the content on the screen at any given time. If that was what the original post was about, we'd be done. It wasn't. Here's the original thread: " We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a "hotdesk" type system where anyone can log on to an extension - however what I would love to do is relabel the phone with the current "owner" when this logon happens. I know that I can change the sip.conf and phones tftp file, however this is a big problem with the Cisco's as they take *forever* (ok, maybe 2 / 3 minutes) to reboot (VLAN problem) 1) Has anyone actually solved this VLAN issue with the cisco ? 2) Is there any way of changing a label without rebooting the phone ? TIA Julian" So for the archives, the prescribed method from Cisco to change the phone's configuration (e.g. the label) without causing the phone to reboot [and take 3 minutes of productivity away from an employee] is to issue an "erase protflash" from the phone's telnet console. This will allow the OP to change the label for the phone [line] for the end user without the painful reboot. When a Cisco phone is running the SIP firmware, it erases the current SIP configuration and requests the configuration files from the TFTP server. It does it rather quickly (at roughly 11 seconds). You must have the telnet_level set to 2 and a phone_password already set in the configuration loaded on the phone. If not, update the configuration files to on the TFTP server (either SipDefault.cnf or the SIP.cnf file) to set these settings and reboot the phone. As long as the telnet_level is set to 2 and a phone_password is set and loaded into the phone's active configuration, you should be able set to update the configuration remotely without the need to reboot the phone. Now - can we get more information about the VLAN issue you're having? -Elliot -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jacek Blaschke Sent: Monday, November 16, 2009 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ODP: Re: Changing labels on Phones It is not the VLAN problem. Simply reboot of the 79xx takes up to 3 minutes and we found no way to speed it up (phones works with Call Manager and VLAN's are not implemented). There are some other methods to display content on the phone screen without editing local configs. Check http://www.ciptec.co.uk/ - commercial site but shows the way. Jacek - Wiadomość oryginalna - Od:: Elliot Otchet Data:: Poniedziałek, 16 Listopad 2009 17:35 Temat: Re: [asterisk-users] Changing labels on Phones > If you just want to change the SIP configuration on the phone > remotely there are a few prerequisites: > > 1) Upgrade your phone's firmware to a recent release. > In the phone's config file on the tftp server: > 2) Set telnet_level set to 2. Make sure it stays at 2 when you > create your new config file for the phone, otherwise you won't be > able to redo this via telnet. > 3) You need a phone_password set. > If the phone was not loaded with the telnet_level at 2, you will > need to reboot the phone manually first for this method to work. > > With those two items out of the way, you (or your script/program) > can telnet to the phone and issue an "erase protflash" which will > cause the phone to erase the current SIP configuration and re-read > the SIP config files only - without the painful reboot. Takes > about 11 seconds on my old 7960 to do. > > There are some security risk associated with the above, but if you > understand them and live with them or can mitigate them in other > ways, you're set. > > -Elliot > > -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk- > users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman > Sent: Monday, November 16, 2009 10:20 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Changing labels on Phones > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian > > > > We have several types of phones, Cisco 79xx, Aastra 9133i etc. > We
Re: [asterisk-users] Odd Local Channel and 0 billsec issue
On Monday 16 November 2009 04:01:22 am Ishfaq Malik wrote: > I've been noticing an odd issue with our servers (1.4.17) where a large > number of one particular customer's (we operate a hosted VoIP platform) > calls go through a Local channel rather than the SIP channel and > whenever this happens our asterisk CDR is recording a billsec value of 0. Most likely cause is that somebody has set up their phone to forward to another extension. When that happens, Dial accomplishes the redirect by generating a Local channel in place of the original. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with sounds DTMF's phone keys
Hello everybody, I need help, I have a problem with conferences in asterisk, when many people are in a conference sometimes there're users pressing phone keys and this action emits a sound (DTMF of the phone keys), so, I need to find the way of not listening this sound.. I'm using MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because users continue listening de DTMF's sounds... But the way, I'm using Asterisk 1.4.26.1 Thanks a lot. -- Saludos cordiales, Diana López ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] can't call through voip provider
Hello.
Sorry to repost this message but, I don't have the original message in my inbox
nor in my sent box.
Well, last week I posted a problem I am having trying to use an asterisk server
use a voip provider and a pstn. Pstn works fine but, I cant even connect to my
provider's server. I don't know what I'm doing wrong.
I tried using a soft phone and I'm able to register and make calls with it but,
when it comes to rerouting the call through asterisk I not able to establish a
call.
This is my setup:
modem -- router/firewall LAN
The asterisk server is on the lan side. I have the modem in bridge mode which
assings my router/firewall the external ip address. I have FORWARD to ACCEPT
in the router and I still cant establish a connection.
My sip.conf file looks like this:
[general]
externhost=optimumwireless.com
localnet=172.16.0.0/16
register => username:sec...@my.service_provider.tld
language=es
;allow=gsm
allow=all
[voipprovider]
type=friend
host=208.78.163.3
username=username
fromuser=username
secret=password
port=5060
dtmfmode=rfc2833
nat=yes
insucure=port,invite
allow=all
careinvite=yes
I don't know what else to try. When I try to call I get this at the cli:
== Using SIP RTP CoS mark 5
-- Executing [91xxx763x...@default:1] Dial("SIP/102-b6a06a40",
"SIP/1xxx763x...@voipprovider") in new stack
== Using SIP RTP CoS mark 5
-- Called 1xxx763x...@voipprovider
Please help me with this I'm running out of options.
Thanks in advanced for your help.
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Re: [asterisk-users] Problem with sounds DTMF's phone keys
On Mon, Nov 16, 2009 at 3:27 PM, Diana Lopez wrote: > Hello everybody, > > I need help, I have a problem with conferences in asterisk, when many > people are in a conference sometimes there're users pressing phone keys > and this action emits a sound (DTMF of the phone keys), so, I need to > find the way of not listening this sound.. I'm using > MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because > users continue listening de DTMF's sounds... > > But the way, I'm using Asterisk 1.4.26.1 > > Thanks a lot. > This is just a guess, but, I think that if any of the users is using inband DTMF and its sip.conf is not enabled to detect inband dtmf, you may have this result. The 'F' options MUST work if all the peers are properly configured, otherwise is a bug. I order to asterisk to mute or ignore that DTMF it must first properly detect it. Use the /etc/asterisk/logger.conf configuration to display detected dtmf and make sure all users dtmf is being detected. If the dtmf is detected by asterisk and you use the F option in meetme and still you hear the dtmf, then is a bug, try to reproduce it with latest 1.4 version, if you can reproduce it the bug must be filled in issues.asterisk.org. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Limit IAX calls on a peer, in and out
We have setup an * box for a small client with 10 phones. They have a 4500/500k ADSL connection which works great when no more than 8 external calls are in progress. (ulaw) The problem is when all 10 people try to use an external channel, AND/OR, 8+ incoming calls arrive at once. The symptom when this happens is obviosuly dropped calls or choppy calls. Of course the ADSL connection is a problem, but I must first deal with the call drops/choppiness before the client invests in a bigger pipe. (loss of client confidence). Is there a call-limit like in sip.conf? I can dream up a way to limit outbound calls in my dial macro...but that gets messy and doesn't deal withinbound. IDEAS? Thanks, MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with sounds DTMF's phone keys
Could it be your using option X when you have no extensions for the user to exit to (therefore when they press dtmf instead of one and done, they just keep going?) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Moises Silva Sent: Monday, November 16, 2009 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with sounds DTMF's phone keys On Mon, Nov 16, 2009 at 3:27 PM, Diana Lopez wrote: Hello everybody, I need help, I have a problem with conferences in asterisk, when many people are in a conference sometimes there're users pressing phone keys and this action emits a sound (DTMF of the phone keys), so, I need to find the way of not listening this sound.. I'm using MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because users continue listening de DTMF's sounds... But the way, I'm using Asterisk 1.4.26.1 Thanks a lot. This is just a guess, but, I think that if any of the users is using inband DTMF and its sip.conf is not enabled to detect inband dtmf, you may have this result. The 'F' options MUST work if all the peers are properly configured, otherwise is a bug. I order to asterisk to mute or ignore that DTMF it must first properly detect it. Use the /etc/asterisk/logger.conf configuration to display detected dtmf and make sure all users dtmf is being detected. If the dtmf is detected by asterisk and you use the F option in meetme and still you hear the dtmf, then is a bug, try to reproduce it with latest 1.4 version, if you can reproduce it the bug must be filled in issues.asterisk.org. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Change canreinvite=yes/no from dialplan?
Hi All,
Currently I have voice calls from a certain SIP peer coming into an asterisk
server where the specific [SIP] channel is set to 'canreinvite=no'.
I would like to enable reinvites for certain calls, matched on DID. So I'm
wondering if there is a mechanism in the dial plan to turn on/off reinvite
capability or will every call on this channel be forced to use the SIP peer
context for the duration of the call? Is there maybe a new feature in 1.6
that does this?
exten => 5551212,1,Set(canreinvite=yes)
exten => 5551212,2,Dial(SIP/${ext...@othersippeer
,,)
Something like that.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
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[asterisk-users] Queues
Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
It should be realistic, but have you considered just using followme to add the cell phones to the queue list? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pbx-cards
Do i rely need a pbx-card to use astersik for voicemail-system? Or can i only use my internal pci-modem A-link ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: "Danny Nicholas" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
Since followme is "extension-based", you have at least two options. Option 1 is to have a few extensions designated for "following" where you punch in the cell numbers as you wish. Option 2 is to use "day logic" to point to the "following" guys based on days.If I were doing option 2, I'd try to use ASTDB to control this instead of having to code a lot of dialplan, but that's just me. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: "Danny Nicholas" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
Hi Travis, There's lots of different ways to attack "on-call" roster solutions in Asterisk - as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn't always suit the "business need". However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his "on-call roster" routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list - they could simply select the right person, and click a "First Call" button to make that person the first in the roster, select another person and click a "Second Call" button to make that person the second in the roster, and so on. Using the Asterisk manager interface - (or even "asterisk -rx " if you're not comfortable using the AMI) - you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk - there is more than one way to "hump the leg". Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is "extension-based", you have at least two options. Option 1 is to have a few extensions designated for "following" where you punch in the cell numbers as you wish. Option 2 is to use "day logic" to point to the "following" guys based on days.If I were doing option 2, I'd try to use ASTDB to control this instead of having to code a lot of dialplan, but that's just me... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: "Danny Nicholas" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design & Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CD
Re: [asterisk-users] Pbx-cards
mattias wrote: > Do i rely need a pbx-card to use astersik for voicemail-system? > Or can i only use my internal pci-modem If you want to connect to an analog phone line, then you will need to use some sort of hardware to connect it (no, a PCI modem is not possible*) -- this can either be a PCI card, or simply an analog-to-SIP converter made by various companies such as Linksys, etc... Alternatively, you can use a SIP provider, and there are many, to get calls into and out of your system. For the cost of a piece of hardware, you're likely to get anywhere from a couple of months to several months of service. Some of which allow you to use it like a pre-paid phone card, and is around a penny per minute. [* Yes, you can use a certain PCI modem which contains the appropriate chipset (not any PCI modem), but the time and money to get such a modem is not worth the time.] Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk VoIP Security Webinar - Video Now Available
The full VoIP Security webinar from last Friday is now available on Asterisk.org http://www.asterisk.org/security/webinar Thanks, -S Steven Sokol Digium, Inc. | Marketing Director - Asterisk 1568 South Yorktown Place – Tulsa, OK – 74104 direct: +1 256-428-6101 mobile: +1 816-806-8844 fax: +1 816-817-0441 twitter: ssokol | jabber: sso...@digium.com | skype: ssokol.digium Check us out at www.asterisk.org & www.digium.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
Hi Michael, Your web interface for the "on-call roster" is pretty close to what we're trying to trying to achieve. I would like to have people signing into the on-call queue be the method that determined whose cell phone to call. I was hoping there was a way to pass the call exiting the queue to a variable or two that was composed of the extensions currently logged into the queue. I set up an extension number for people to call into and enter a forwarding number which writes an entry into the ASTDB. I have my dialplan check to see if there is an ASTDB entry for that extension before it tries to dial their deskphone, and if there is an entry it dials the forwarded number stored in the database instead. The closest thing so far I have found to what I am trying to achieve is to hard code a couple of spare extensions into the dialplan, and then have whoever is on-call set one of those extensions to their cell phone number. I'll definitely take another look at followme to see if I can adapt that what I'm trying to achieve. Thanks Danny & Michael, Travis - Original Message - From: "Michael Wyres" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 2:23:49 PM Subject: Re: [asterisk-users] Queues Hi Travis, There’s lots of different ways to attack “on-call” roster solutions in Asterisk – as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn’t always suit the “business need”. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his “on-call roster” routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list – they could simply select the right person, and click a “First Call” button to make that person the first in the roster, select another person and click a “Second Call” button to make that person the second in the roster, and so on. Using the Asterisk manager interface – (or even “asterisk –rx ” if you’re not comfortable using the AMI) – you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk – there is more than one way to “hump the leg”. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is “extension-based”, you have at least two options. Option 1 is to have a few extensions designated for “following” where you punch in the cell numbers as you wish. Option 2 is to use “day logic” to point to the “following” guys based on days. If I were doing option 2, I’d try to use ASTDB to control this instead of having to code a lot of dialplan, but that’s just me… From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: "Danny Nicholas" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an "on-call" queue. A call comes in and it rings the "on-call" extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the "on-call" queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueue
[asterisk-users] Question about OSLEC or HPEC with AsteriskNow
I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my previous config to the new computer. Now everything is fine except that even if I use the md2 echo cancellation it's not perfect I still have echo issue. So I made some search around and found that there's oslec and hpec out there that seems to be better then what I'm currently using. So my question should I use hpec or oslec with my TDM400 card? I also tried to recompile dahdi to use oslec (before I found that Digium had hpec) but then I get an error message that the source of my kernel cannot be found so I can never actually compile a new version of dahdi. Thanks Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Understanding Congestion to incoming caller
I have an * installation which will refuse incoming callers once a max (5 callers) is reached. Caller 6 and up should be notified of congestion...without network load on my trunk. How would I do this? The voipinfo wiki shows playing a congestion tone to the caller, but that seems stupid since I'm consuming bandwidth to send a tone. I also tried just responding with the congestion command, but the 6th+ call just hears a hangup when calling in. Can someone explain how this should be done? Thanks, MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues
Again – lots of ways to do it – you could use a web interface to set the numbers in ASTDB for lookup – or you could create an IVR to ask for the number, and store it in ASTDB that way. Good luck! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Tuesday, 17 November 2009 10:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues Hi Michael, Your web interface for the "on-call roster" is pretty close to what we're trying to trying to achieve. I would like to have people signing into the on-call queue be the method that determined whose cell phone to call. I was hoping there was a way to pass the call exiting the queue to a variable or two that was composed of the extensions currently logged into the queue. I set up an extension number for people to call into and enter a forwarding number which writes an entry into the ASTDB. I have my dialplan check to see if there is an ASTDB entry for that extension before it tries to dial their deskphone, and if there is an entry it dials the forwarded number stored in the database instead. The closest thing so far I have found to what I am trying to achieve is to hard code a couple of spare extensions into the dialplan, and then have whoever is on-call set one of those extensions to their cell phone number. I'll definitely take another look at followme to see if I can adapt that what I'm trying to achieve. Thanks Danny & Michael, Travis - Original Message - From: "Michael Wyres" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 2:23:49 PM Subject: Re: [asterisk-users] Queues Hi Travis, There’s lots of different ways to attack “on-call” roster solutions in Asterisk – as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn’t always suit the “business need”. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his “on-call roster” routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list – they could simply select the right person, and click a “First Call” button to make that person the first in the roster, select another person and click a “Second Call” button to make that person the second in the roster, and so on. Using the Asterisk manager interface – (or even “asterisk –rx ” if you’re not comfortable using the AMI) – you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk – there is more than one way to “hump the leg”. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is “extension-based”, you have at least two options. Option 1 is to have a few extensions designated for “following” where you punch in the cell numbers as you wish. Option 2 is to use “day logic” to point to the “following” guys based on days.If I were doing option 2, I’d try to use ASTDB to control this instead of having to code a lot of dialplan, but that’s just me… From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: "Danny Nicholas" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone,
Re: [asterisk-users] can't call through voip provider
On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy wrote:
> I don't know what else to try. When I try to call I get this at the cli:
>
> == Using SIP RTP CoS mark 5
> -- Executing [91xxx763x...@default:1] Dial("SIP/102-b6a06a40",
> "SIP/1xxx763x...@voipprovider") in new stack
> == Using SIP RTP CoS mark 5
> -- Called 1xxx763x...@voipprovider
>
We could really use a little more of the CLI output of a failed call. Maybe
increase your verbosity to at least 10. Also, what does the SIP debug of a
call to the VOIP provider look like (from the cli, type "sip set debug peer
voipprovider")?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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[asterisk-users] max call duration
How can I set a maximum call duration on a ZAP channel? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7971 behind NAT
Hi all, does anyone have any luck using a Cisco 7971 (SIP) behind NAT with two different accounts on the same server (i.e. two different extensions)? I am using Cisco-CP7971G-GE/8.3.0 and asterisk V1.4.something. The phone sends SIP packets from a high-numbered UDP port but expects a reply on port 5060. Fine, I do some magic with iptables to rewrite the packets (which limits me to one phone at that location, unless I'm mistaken). Incoming calls work fine on both accounts, but outgoing calls work only from the most recently registered account (the order is random due to timing) since both appear to asterisk as IP:5060. An outgoing call from the other account is rejected with an authentication mismatch, which makes sense. Asterisk matches the most recently registered peer by IP/port and if the user name differs, it complains, even if the password is the same for both accounts. So, is this the worst SIP implementation ever in those Cisco 7971's or am I doing something very wrong here? Technically even without NAT this confusion would occur as both accounts use IP:5060 so Asterisk cannot tell them apart during the initial peer matching stage. Of course the source port the Cisco selects is different with every dialog, so that doesn't help either. Any input would be appreciated before I throw that phone out of the window. Thanks, Luki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7971 behind NAT
You need to enable SIP transformations on the firewall, the packets will have to be dynamically re-written to handle multiple Cisco phones of these models. Be sure 'nat=no' is set in sip.conf for the phones as well, or Asterisk will reply to the incorrect ports (source instead of the mangled contact header). In this case, you'll need to compile in the SIP connection tracking/NAT bits in the kernel, they should be able to mangle the packets appropriately. I have never tested this, as all my deployments have hardware firewalls with SIP support built-in. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luki Sent: Monday, November 16, 2009 20:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7971 behind NAT Hi all, does anyone have any luck using a Cisco 7971 (SIP) behind NAT with two different accounts on the same server (i.e. two different extensions)? I am using Cisco-CP7971G-GE/8.3.0 and asterisk V1.4.something. The phone sends SIP packets from a high-numbered UDP port but expects a reply on port 5060. Fine, I do some magic with iptables to rewrite the packets (which limits me to one phone at that location, unless I'm mistaken). Incoming calls work fine on both accounts, but outgoing calls work only from the most recently registered account (the order is random due to timing) since both appear to asterisk as IP:5060. An outgoing call from the other account is rejected with an authentication mismatch, which makes sense. Asterisk matches the most recently registered peer by IP/port and if the user name differs, it complains, even if the password is the same for both accounts. So, is this the worst SIP implementation ever in those Cisco 7971's or am I doing something very wrong here? Technically even without NAT this confusion would occur as both accounts use IP:5060 so Asterisk cannot tell them apart during the initial peer matching stage. Of course the source port the Cisco selects is different with every dialog, so that doesn't help either. Any input would be appreciated before I throw that phone out of the window. Thanks, Luki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7971 behind NAT
Darryl, OK, that could work but it makes the use of these phones behind consumer routers rather impossible. How many of those will inspect and transform SIP packets? Oh why does Cisco have to do things differently from everyone else... Luki 2009/11/16 Darryl Dunkin : > You need to enable SIP transformations on the firewall, the packets will > have to be dynamically re-written to handle multiple Cisco phones of > these models. Be sure 'nat=no' is set in sip.conf for the phones as > well, or Asterisk will reply to the incorrect ports (source instead of > the mangled contact header). > > In this case, you'll need to compile in the SIP connection tracking/NAT > bits in the kernel, they should be able to mangle the packets > appropriately. I have never tested this, as all my deployments have > hardware firewalls with SIP support built-in. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7971 behind NAT
On Mon, Nov 16, 2009 at 10:53 PM, Luki wrote: > Darryl, > > OK, that could work but it makes the use of these phones behind > consumer routers rather impossible. How many of those will inspect and > transform SIP packets? Oh why does Cisco have to do things differently > from everyone else... > > Luki > > 2009/11/16 Darryl Dunkin : > > You need to enable SIP transformations on the firewall, the packets will > > have to be dynamically re-written to handle multiple Cisco phones of > > these models. Be sure 'nat=no' is set in sip.conf for the phones as > > well, or Asterisk will reply to the incorrect ports (source instead of > > the mangled contact header). > > > > In this case, you'll need to compile in the SIP connection tracking/NAT > > bits in the kernel, they should be able to mangle the packets > > appropriately. I have never tested this, as all my deployments have > > hardware firewalls with SIP support built-in. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > I use two accounts on a Cisco 7941 at home that is connected to my asterisk server running at a datacenter. My home has NAT, my asterisk server does not. I do not need to do any of the packet mangling stuff, just set "nat=no" in the sip.conf entry for the Cisco phone. Not sure how much different the 7971 is though... -- Thanks, --Warren Selby http://www.selbytech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pbx-cards
But are not pbx card and modem the same? -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Leif Madsen Skickat: den 16 november 2009 23:46 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Pbx-cards mattias wrote: > Do i rely need a pbx-card to use astersik for voicemail-system? Or can > i only use my internal pci-modem If you want to connect to an analog phone line, then you will need to use some sort of hardware to connect it (no, a PCI modem is not possible*) -- this can either be a PCI card, or simply an analog-to-SIP converter made by various companies such as Linksys, etc... Alternatively, you can use a SIP provider, and there are many, to get calls into and out of your system. For the cost of a piece of hardware, you're likely to get anywhere from a couple of months to several months of service. Some of which allow you to use it like a pre-paid phone card, and is around a penny per minute. [* Yes, you can use a certain PCI modem which contains the appropriate chipset (not any PCI modem), but the time and money to get such a modem is not worth the time.] Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about OSLEC or HPEC with AsteriskNow
2009/11/17 Martin Roy > I was previously using an old computer running Asterisk 1.2 with > zaptel. Once the CPU fried I switch to a new computer and I chose > AsteriskNow 1.5 running in 64bits to simplify the installation > process. I manage to find my way with configuring dahdi instead of > zaptel and to switch all my previous config to the new computer. Now > everything is fine except that even if I use the md2 echo cancellation > it's not perfect I still have echo issue. So I made some search around > and found that there's oslec and hpec out there that seems to be > better then what I'm currently using. So my question should I use hpec > or oslec with my TDM400 card? I also tried to recompile dahdi to use > oslec (before I found that Digium had hpec) but then I get an error > message that the source of my kernel cannot be found Do you imply you previously installed a Dahdi binary package ? If positive, before compiling Dahdi source code, you need to install Linux header files. On Debian systems, you can get this with something like : apt-get -install linux-headers-2.6.26-2-686 Regards > so I can never > actually compile a new version of dahdi. > > Thanks > > Martin > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Congestion to incoming caller
2009/11/17 Michelle Dupuis > I have an * installation which will refuse incoming callers once a max (5 > callers) is reached. Caller 6 and up should be notified of > congestion...without network load on my trunk. > On which tech, does this trunk rely ? Is it a SIP trunk ? > How would I do this? > > The voipinfo wiki shows playing a congestion tone to the caller, but that > seems stupid since I'm consuming bandwidth to send a tone. > > I also tried just responding with the congestion command, but the 6th+ call > just hears a hangup when calling in. > > Can someone explain how this should be done? > > Thanks, > MD > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
