Re: [asterisk-users] Polycom retrieve call from hold
On Saturday 28 November 2009 05:05:47 pm Kevin P. Fleming wrote: > Darrick Hartman wrote: > >> The phone is a Polycom 501; it's been discontinued. I am working on a > >> testing/migration plan to move to the latest Asterisk 1.6.x, but I'm > >> hesitant to upgrade a system that doesn't currently work right. > > There's no particular reason that you need to move to 1.6.x, but > certainly running something newer from the 1.4.x series than 1.4.0-beta2 > would be preferred and much easier to support. Well, as you can tell, I don't upgrade very often. When I can/do, I need to get as much bang for my buck as I can. -- Take care and have fun, Mike Diehl. ___ -- 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] Gtalk Asterisk integration
Hello users,
I am trying to integrate asterisk and gtalk.
my configuration is as follows
OS:centos
asterisk-1.6.0
asterisk-addons-1.6.0
dahdi-linux-2.2
dahdi-tools-2.2
libpri-1.4 share
iksemel-1.2
#/etc/asterisk/jabber.conf
[general]
debug=yes
autoprune=no
autoregister=no
[google]
type=client
serverhost=talk.google.com
username=x...@gmail.com
secret=x
port=5222
usetls=yes
usesasl=yes
statusmessage="Invox Google Talk"
timeout=100
# /etc/asterisk/gtalk.conf[general]
context=google-in
bindaddr=192.168.1.74
allowguest=yes
[guest]
;disallow=all
allow=ulaw
context=google-in
[test]
username=y...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=google
# /etc/asterisk/extensions.conf
[google-in]
;Incoming
exten => s,1,NoOp(call from Google Talk)
exten => s,n,Set(CALLERID(name)="From Google Talk")
exten => s,n,Dial(SIP/1000,20,r)
exten => s,n,Hangup()
;Outgoing
exten => 100,1,JabberStatus(google,y...@gmail.com,STATUS)
exten => 100,n,NoOp(Jabber Status=${STATUS})
exten => 100,n,Dial(Gtalk/google/invoxgt...@gmail.com/Talk)
exten => 100,n,Hangup()
# /etc/asterisk/rtp.conf
rtpstart=1650
rtpend=4560
ports opened on the router
tcp 443 -incoming, outgoing
tcp 5222-incoming,outgoing
udp- all open incoming, outgoing
-> i am able to call from my external gtalk client to the server configured
user
# this case is working fine
Executing [...@google-in:1] NoOp("Gtalk/Y-49af", "call from Google
Talk") in new stack
Executing [...@google-in:2] Set("Gtalk/Y-49af", "CALLERID(name)="From
Google Talk"") in new stack
Executing [...@google-in:3] Dial("Gtalk/Y-49af", "SIP/1000,20,r") in new
stack
Using SIP RTP CoS mark 5
-- Called 1000
-> when i try to call from asterisk to the external client
# this case is not working and throwing following error
Executing [...@google-in:1] JabberStatus("SIP/1000-000e", "google,
yy...@gmail.com,STATUS") in new stack
Executing [...@google-in:2] NoOp("SIP/1000-000e", "Jabber
Status=1") in new stack
Executing [...@google-in:3] Dial("SIP/1000-000e", "Gtalk/google/
yy...@gmail.com/Talk") in new stack
[Nov 30 16:22:25] ERROR[16255]: chan_gtalk.c:932 gtalk_alloc: no gtalk
capable clients to talk to.
[Nov 30 16:22:25] WARNING[16255]: app_dial.c:1518 dial_exec_full:
Unable to create channel of type 'Gtalk' (cause 0 - Unknown)
Everyone is busy/congested at this time (1:0/0/1)
Executing [...@google-in:4] Hangup("SIP/1000-000e", "") in new
stack
Spawn extension (google-in, 100, 4) exited non-zero on
'SIP/1000-000e'
###
asterisk cli out put
###
> jabber show connected
User: xx...@gmail.com - Connected
>jabber show buddies
yy...@gmail.com
localhost*CLI> Resource: Talk.v104C77D0BCE
localhost*CLI> node: http://www.google.com/xmpp/client/caps
localhost*CLI> version: 1.0.0.104
localhost*CLI> Jingle capable: yes
> gtalk show channels
Channel Jabber ID Resource
Read Write
0 active gtalk channels
i am getting error "no gtalk capable clients to talk to"
i tried with both asterisk 1.4.25 version and asterisk 1.6.0 but no
difference
anybody can help me out finding a forward move on this???
Thanks in advance
srinivas
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Re: [asterisk-users] AGI stuff
Hallo Philipp, Wei Gehts ist Einen. Danke. I am in USA. Thanks. On Sun, Nov 29, 2009 at 8:49 PM, Philipp Kempgen wrote: > Thomas Perron schrieb: >> How do I get to this prompt? >> >> #!/usr/bin/php -q >> > http://en.wikipedia.org/wiki/Shebang_%28Unix%29 > > > Philipp Kempgen > -- > AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de > Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de > -- > > ___ > -- 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] AGI stuff
Thomas Perron schrieb: > How do I get to this prompt? > > #!/usr/bin/php -q > http://en.wikipedia.org/wiki/Shebang_%28Unix%29 Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- 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] AGI stuff
How do I get to this prompt? #!/usr/bin/php -q 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] Max how many users in sip.conf
On Sunday 29 November 2009 17:03:04 Leif Neland wrote: > mtha...@gmail.com skrev: > > Anyone know how many users i can record in sip.conf. (NO..NO i am not > > discussing the simultaneous sip calls). > > I tried with 50k users in sip.conf, but the sip module didn't reload. > > tried with few hundred of users and it works. any idea what is the > > limit in sip.conf > > Try a binary search > in 15 tries you have the exact value. > > Start with 32768 entries. > If it works, add 32768/2 =16384. > If not, subtract 16384, giving 16384. > > Continue, adding/subtracting > 8192,4096.2048,1024,512,256,128,64,32,16.8,4,2,1 There is no maximum. However, if you have a typo in there somewhere, the entire file will fail to load. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Max how many users in sip.conf
mtha...@gmail.com skrev: > > Anyone know how many users i can record in sip.conf. (NO..NO i am not > discussing the simultaneous sip calls). > I tried with 50k users in sip.conf, but the sip module didn't reload. > tried with few hundred of users and it works. any idea what is the > limit in sip.conf Try a binary search in 15 tries you have the exact value. Start with 32768 entries. If it works, add 32768/2 =16384. If not, subtract 16384, giving 16384. Continue, adding/subtracting 8192,4096.2048,1024,512,256,128,64,32,16.8,4,2,1 Leif ___ -- 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 H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)
bilal ghayyad schrieb: > To be able run Asterisk and gnugk on the same machine at same IP address, I > need to know how to configure the port ranges of the (Q931, H245, T120, RTP) > for the asterisk H323 channel to avoid any confilict with the gnugk? From > where to determine these ranges? > > About gnugk, I know from where to determine it, but I do not know how to > determine these port ranges in the Asterisk H323. Not really an answer to your question but why not simply use different IP addresses? (bindaddr) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ___ -- 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 H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)
Hi All; I am wondering of this H323 channel in asterisk, whatever I ask, I do not get help :) - So, how to get help, I do not know. To be able run Asterisk and gnugk on the same machine at same IP address, I need to know how to configure the port ranges of the (Q931, H245, T120, RTP) for the asterisk H323 channel to avoid any confilict with the gnugk? From where to determine these ranges? About gnugk, I know from where to determine it, but I do not know how to determine these port ranges in the Asterisk H323. Any help? Regards Bilal ___ -- 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] Parsing custom SIP headers
Hi,
Just to be sure: Is there a dialplan function in Asterisk that
parses custom "name-addr"-style SIP headers for me?
If I wanted to do it right the syntax
name-addr *(SEMI generic-param)
is quite complex to parse in the dialplan using nothing but CUT().
It's so easy to make false assumtions about angle brackets (< >),
whitespace (LWS), quotes (") around the display-name, character
escaping etc. All of the applications of CUT() I have seen are
way too simplistic.
Example of how it could work:
Set(addr=${SIP_PARSE_HEADER(${SIP_HEADER(P-Asserted-Identity)},addr-spec)});
Interesting parts include:
name-addr, display-name, addr-spec, scheme, userinfo, user,
telephone-subscriber, host, hostname, port, ...
Actually headers like P-Asserted-Identity can even have more then
one value.
---cut---
PAssertedID = "P-Asserted-Identity" HCOLON PAssertedID-value
*(COMMA PAssertedID-value)
PAssertedID-value = name-addr / addr-spec
---cut---
so I guess SIP_PARSE_HEADER() would need an index argument, just
like SIP_HEADER().
Proper parsing could be done in an AGI() script of course but that
involves a big overhead especially since the code to parse name-addr
is already in Asterisk. It's just not available in the dialplan.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de
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Re: [asterisk-users] Portec - feedback wanted
On Mon, 23 Nov 2009 09:21:23 Michael wrote: > On Mon, 23 Nov 2009 08:54:34 F6HQZ wrote: > > Hi Michael, > > > > It does what it is announced/supposed to do. > > I have checked and know well all the Portech GSM/SIP family. > > > > But, be carefull, because under the same reference you can buy/receive > > different hardware versions : - 2, 3 or 4 GSM frequencies bands > > - Siemens or Simcom GSM modules Ok, I want to buy a quad band MV370 with Siemens radio. If someone on this list has this for sale (new or used) they should get in contact with me without delay. Michael ___ -- 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
