Re: [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions
Hmm,
First i must correct myself, MEMBERINTERFACE seems to be NULL, not the “device”
that called in, my bad reading.
Did some changes:
queues.conf
---
[Kinna]
keepstats=yes
ringinuse=no
setinterfacevar=yes
setqueuevar=yes
strategy=rrmemory
timeout=5
wrapuptime=120
extensions.conf
exten = 0320209030,1,Answer()
exten =
0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT))
exten = 0320209030,n,Queue(Kinna,nrtT)
exten = 0320209030,n,NoOp(${MEMBERINTERFACE})
exten = 0320209030,n,NoOp(${MEMBERNAME})
exten = 0320209030,n,NoOp(${QUEUENAME})
exten = 0320209030,n,Queue(Goteborg,rtT)
exten = 0320209030,n,Hangup()
Same call flows as below:
== Using SIP RTP CoS mark 5
-- Executing [0320209...@inputinterior.se:1]
Answer(SIP/0317998985-0050, ) in new stack
-- Executing [0320209...@inputinterior.se:2]
ExecIf(SIP/0317998985-0050, 0?Queue(Goteborg,rtT)) in new stack
-- Executing [0320209...@inputinterior.se:3] Queue(SIP/0317998985-0050,
Kinna,nrtT) in new stack
== Using SIP RTP CoS mark 5
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- Nobody picked up in 5000 ms
-- Exiting on time-out cycle
-- Executing [0320209...@inputinterior.se:4] NoOp(SIP/0317998985-0050,
) in new stack
-- Executing [0320209...@inputinterior.se:5] NoOp(SIP/0317998985-0050,
) in new stack
-- Executing [0320209...@inputinterior.se:6] NoOp(SIP/0317998985-0050,
Kinna) in new stack
-- Executing [0320209...@inputinterior.se:7] Queue(SIP/0317998985-0050,
Goteborg,rtT) in new stack
QUEUENAME is working the way i am excpecting but MEMBERINTERFACE and MEMBERNAME
is not, or am I wrong?
From: magnu...@inputinterior.se
Sent: Sunday, February 20, 2011 8:05 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions
Hi!
Did play around with queues and need some help. I thought that MEMBERINTERFACE
and MEMBERNAME should be set to the “device” the call was queued to not the
device that called the queue, or do i miss something?
Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on
2011-01-31 13:38:23 UTC
0317998985 calls Kinna (0320209030)
Tomas Ekman (SIP/0317998972) receives the call but don’t answer.
When the queue “timeout” I would like to get the name of the device that didn’t
answered, in my case: SIP/0317998972.
${MEMBERINTERFACE} gives me the name of the device that called in.
queue show Kinna
Kinna has 0 calls (max unlimited) in 'rrmemory' strategy (4s holdtime, 2s
talktime), W:0, C:1, A:13, SL:0.0% within 0s
Members:
Tomas Ekman (SIP/0317998972) with penalty 1 (dynamic) (Not in use) has
taken no calls yet
No Callers
queues.conf
---
[general]
;
autofill=yes
keepstats=yes
setinterfacevar=yes
;
[Kinna]
retry=5
ringinuse=no
strategy=rrmemory
timeout=20
wrapuptime=120
extensions.conf
---
exten = 0320209030,1,Answer()
exten =
0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT))
exten = 0320209030,n,Queue(Kinna,nrtT)
exten = 0320209030,n,NoOp(${MEMBERINTERFACE})
exten = 0320209030,n,NoOp(${MEMBERNAME})
exten = 0320209030,n,Queue(Goteborg,rtT)
exten = 0320209030,n,Hangup()
CLI
== Using SIP RTP CoS mark 5
-- Executing [0320209...@inputinterior.se:1]
Answer(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:2]
ExecIf(SIP/0317998985-0033, 0?Queue(Goteborg,rtT)) in new stack
-- Executing [0320209...@inputinterior.se:3]
Queue(SIP/0317998985-0033, Kinna,nrtT) in new stack
== Using SIP RTP CoS mark 5
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
-- Executing [0320209...@inputinterior.se:4]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:5]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:6]
Queue(SIP/0317998985-0033, Goteborg,rtT) in new stack
== Spawn extension (inputinterior.se, 0320209030, 6) exited non-zero on
'SIP/0317998985-0033'
Could any help me understand what I am doing wrong?
/Magnus
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Re: [asterisk-users] DTMF and Snom
Jonas Kellens jonas.kell...@telenet.be writes: Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. Which phone model? If 870, you may want to look at this thread: http://forum.snom.com/index.php?showtopic=4084 You may want to experiment with a different firmware version. It is a bit surprising that they do not allow you to set the DTMF duration and volume. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meet me recording
Does it create separet file foreach channel? Or single one? -- Sent from my iPhone On Feb 19, 2011, at 12:45 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi Satish, You can Pass 'r' flag to meetme Application and file will be recorded nothin to load mixmonitor and other Application on Channel, i think 'r' is better than all options Cheers Dhaval On Sat, Feb 19, 2011 at 1:37 AM, satish patel satish...@hotmail.com wrote: Thanks, look like monitor application resolved my issue. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Feb 2011 09:16:36 -0600 Subject: Re: [asterisk-users] Meet me recording From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of satish patel Sent: Friday, February 18, 2011 9:12 AM To: asterisk-users Subject: [asterisk-users] Meet me recording Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881- conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881- conf-20110216-130321.wav -rw-r--r-- 1 asterisk asterisk 1604204 Feb 16 13:05 8881- conf-20110216-130337.wav -rw-r--r-- 1 asterisk asterisk 241964 Feb 17 08:20 8881- conf-20110217-081957.wav -rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12 8881- conf-20110217-095056.wav -rw-r--r-- 1 asterisk asterisk 612204 Feb 17 09:53 8881- conf-20110217-095310.wav -rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13 8881- conf-20110217-095414.wav -rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12 8881- conf-20110217-100012.wav -rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12 8881- conf-20110217-100052.wav -rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11 8881- conf-20110217-100117.wav -rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12 8881- conf-20110217-100327.wav -rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06 8881- conf-20110217-102007.wav Thanks, S From what I read, mixmonitor. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote: On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? What version of dahdi do you have installed? I would try using the latest version 2.4.0. It is important to compile and install in the correct order. I usually do dahdi, libpri, asterisk, and then wanpipe. I'm running the latest of everything, except my kernel -- I went with 2.6.32.27 as being a well-maintained long-term kernel. (2.6.37 gave me grief -- too new, I guess.) I'm running -- if it makes a difference -- on an Ubuntu 8.04-4 system. I've re-installed everything, in the order you gave, to, alas, the exact same result: everything seems to initialize, install, etc., correctly, but no dahdi feature in Asterisk. Is there a module I need to load? Or... something? I'd hate to have to revert to 1.4 after all this work. Thanks! -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meet me recording
Satish Patel satish...@hotmail.com wrote: Does it create separet file foreach channel? Or single one? -- Sent from my iPhone On Feb 19, 2011, at 12:45 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi Satish, You can Pass 'r' flag to meetme Application and file will be recorded nothin to load mixmonitor and other Application on Channel, i think 'r' is better than all options Cheers Dhaval On Sat, Feb 19, 2011 at 1:37 AM, satish patel satish...@hotmail.com wrote: Thanks, look like monitor application resolved my issue. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Feb 2011 09:16:36 -0600 Subject: Re: [asterisk-users] Meet me recording From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of satish patel Sent: Friday, February 18, 2011 9:12 AM To: asterisk-users Subject: [asterisk-users] Meet me recording Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881- conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881- conf-20110216-130321.wav -rw-r--r-- 1 asterisk asterisk 1604204 Feb 16 13:05 8881- conf-20110216-130337.wav -rw-r--r-- 1 asterisk asterisk 241964 Feb 17 08:20 8881- conf-20110217-081957.wav -rw-r--r-- 1 asterisk asterisk 78678124 Feb 17 11:12 8881- conf-20110217-095056.wav -rw-r--r-- 1 asterisk asterisk 612204 Feb 17 09:53 8881- conf-20110217-095310.wav -rw-r--r-- 1 asterisk asterisk 81183084 Feb 17 11:13 8881- conf-20110217-095414.wav -rw-r--r-- 1 asterisk asterisk 69488044 Feb 17 11:12 8881- conf-20110217-100012.wav -rw-r--r-- 1 asterisk asterisk 68917164 Feb 17 11:12 8881- conf-20110217-100052.wav -rw-r--r-- 1 asterisk asterisk 66971884 Feb 17 11:11 8881- conf-20110217-100117.wav -rw-r--r-- 1 asterisk asterisk 66648684 Feb 17 11:12 8881- conf-20110217-100327.wav -rw-r--r-- 1 asterisk asterisk 45056044 Feb 17 11:06 8881- conf-20110217-102007.wav Thanks, S From what I read, mixmonitor. It creates just one file for the conference. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote: On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote: On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? What version of dahdi do you have installed? I would try using the latest version 2.4.0. It is important to compile and install in the correct order. I usually do dahdi, libpri, asterisk, and then wanpipe. I'm running the latest of everything, except my kernel -- I went with 2.6.32.27 as being a well-maintained long-term kernel. (2.6.37 gave me grief -- too new, I guess.) I'm running -- if it makes a difference -- on an Ubuntu 8.04-4 system. I've re-installed everything, in the order you gave, to, alas, the exact same result: everything seems to initialize, install, etc., correctly, but no dahdi feature in Asterisk. Is there a module I need to load? Or... something? I'd hate to have to revert to 1.4 after all this work. Thanks! -Ken If you have autoload=yes in modules.conf it should load automatically. Have you checked log, usually /etc/asterisk/full to see if you are getting any error messages relating to dahdi? Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
thanks a lot. that was a problem. On Fri, Feb 18, 2011 at 8:44 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Friday 18 February 2011 05:29:56 Borin wrote: Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Make sure res_ael_share.so is loaded first. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
On Sun, Feb 20, 2011 at 9:44 AM, Ryan Wagoner rswago...@gmail.com wrote: On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote: On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote: On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? What version of dahdi do you have installed? I would try using the latest version 2.4.0. It is important to compile and install in the correct order. I usually do dahdi, libpri, asterisk, and then wanpipe. I'm running the latest of everything, except my kernel -- I went with 2.6.32.27 as being a well-maintained long-term kernel. (2.6.37 gave me grief -- too new, I guess.) I'm running -- if it makes a difference -- on an Ubuntu 8.04-4 system. I've re-installed everything, in the order you gave, to, alas, the exact same result: everything seems to initialize, install, etc., correctly, but no dahdi feature in Asterisk. Is there a module I need to load? Or... something? I'd hate to have to revert to 1.4 after all this work. Thanks! -Ken If you have autoload=yes in modules.conf it should load automatically. Have you checked log, usually /etc/asterisk/full to see if you are getting any error messages relating to dahdi? Ryan You might also do a rm -rf /usr/lib/asterisk/modules/*.so and make install Asterisk again. You could have some modules left around from a previous version conflicting with things. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no progress indication
On 2/18/11 5:18 PM, Paul Belanger pabelan...@digium.com wrote: On 11-02-18 03:59 PM, Cassius Smith wrote: I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). *CLI core show application Progress() CLI output: -- Executing [s@macro-StdExten:6] Dial(IAX2/barneveld-2036, SIP/RickEndpointSIP/xlite-RickEndpoint,20) in new stack == Using SIP RTP CoS mark 5 [Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect [Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument -- Called RickEndpoint [Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument == Spawn extension (macro-StdExten, s, 6) exited non-zero on 'IAX2/barneveld-2036' in macro 'StdExten' == Spawn extension (no911, RickEndpoint, 1) exited non-zero on 'IAX2/barneveld-2036' -- Hungup 'IAX2/barneveld-2036' [Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. There is something going wrong here, netsock2 is not parsing the IP address correctly. Are you using realtime? It would be good to see a full debug[1] log of your call. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Hi Paul, no, not using realtime. I collected the trace but it didn't seem to give much clue (at least to me). Here is an extract from the log (dialing extension 4511 this time). Let me know if you want the full debug log including IAX and SIP debugs. (trunk is IAX, endpoints are SIP). [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Macro' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [4511@no911:1] Macro(IAX2/barneveld-9539, StdExten,SIP/4511SIP/xlite-4511,20) in new stack [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'MACRO_EXTEN' is '4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Verbose' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:1] Verbose(IAX2/barneveld-9539, 2,Processing StdExten call for 4511) in new stack [Feb 20 00:23:23] VERBOSE[9962] app_verbose.c: == Processing StdExten call for 4511 [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Verbose [Feb 20 00:23:23] DEBUG[9962] pbx.c: Function result is 'Cassius Home 3703' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Verbose' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:2] Verbose(IAX2/barneveld-9539, 2,CallerID = Cassius Home 3703) in new stack [Feb 20 00:23:23] VERBOSE[9962] app_verbose.c: == CallerID = Cassius Home 3703 [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Verbose [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG1' is 'SIP/4511SIP/xlite-4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Set' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:3] Set(IAX2/barneveld-9539, Device=SIP/4511SIP/xlite-4511) in new stack [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Set [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'MACRO_EXTEN' is '4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Set' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:4] Set(IAX2/barneveld-9539, UserID=4511) in new stack [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Set [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG1' is 'SIP/4511SIP/xlite-4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG2' is '20' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Dial' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:5] Dial(IAX2/barneveld-9539, SIP/4511SIP/xlite-4511,20) in new stack [Feb 20
Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
On Sat, Feb 19, 2011 at 04:15:15PM -0500, Ken D'Ambrosio wrote: Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? This may be caused by one of two things: 1. You have not built chan_dahdi.so 2. You built chan_dahdi.so, but it has failed to load (normally because of broken configuration) Try running the following from the Asterisk CLI (rasterisk): module unload chan_dahdi.so That one will likely give an error message. Ignore it. module load chan_dahdi.so What error do you get from that? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
You -- as usual -- hit the nail on the head; I'd actually figured it out at probably roughly the same time as you e-mailed, because I bumped into this: Asterisk Module and Build Option Selection [...] XXX chan_dahdi [...] DAHDI Telephony Depends on: dahdi(E), tonezone(E) Can use: res_smdi(M), pri(E), ss7(E), openr2(E) So, I'm using this with a Sangoma A102D; I'm not sure what the E (external?) vs. the M (module?) is about; I've compiled dahdi and tonezone -- how do I verify where the missing dependency lies? Thanks (yet again, all), -Ken On Sun, February 20, 2011 10:38 am, Tzafrir Cohen wrote: On Sat, Feb 19, 2011 at 04:15:15PM -0500, Ken D'Ambrosio wrote: Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? This may be caused by one of two things: 1. You have not built chan_dahdi.so 2. You built chan_dahdi.so, but it has failed to load (normally because of broken configuration) Try running the following from the Asterisk CLI (rasterisk): module unload chan_dahdi.so That one will likely give an error message. Ignore it. module load chan_dahdi.so What error do you get from that? -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions
It's been my experience that the MEMBER... Variables are populated by the
person who answers the queue call. If no one answers the call, I would imagine
the variables would be null.
Thanks,
--Warren Selby, dCAP
On Feb 20, 2011, at 2:17 AM, magnu...@inputinterior.se wrote:
Hmm,
First i must correct myself, MEMBERINTERFACE seems to be NULL, not the
“device” that called in, my bad reading.
Did some changes:
queues.conf
---
[Kinna]
keepstats=yes
ringinuse=no
setinterfacevar=yes
setqueuevar=yes
strategy=rrmemory
timeout=5
wrapuptime=120
extensions.conf
exten = 0320209030,1,Answer()
exten =
0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT))
exten = 0320209030,n,Queue(Kinna,nrtT)
exten = 0320209030,n,NoOp(${MEMBERINTERFACE})
exten = 0320209030,n,NoOp(${MEMBERNAME})
exten = 0320209030,n,NoOp(${QUEUENAME})
exten = 0320209030,n,Queue(Goteborg,rtT)
exten = 0320209030,n,Hangup()
Same call flows as below:
== Using SIP RTP CoS mark 5
-- Executing [0320209...@inputinterior.se:1]
Answer(SIP/0317998985-0050, ) in new stack
-- Executing [0320209...@inputinterior.se:2]
ExecIf(SIP/0317998985-0050, 0?Queue(Goteborg,rtT)) in new stack
-- Executing [0320209...@inputinterior.se:3]
Queue(SIP/0317998985-0050, Kinna,nrtT) in new stack
== Using SIP RTP CoS mark 5
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- Nobody picked up in 5000 ms
-- Exiting on time-out cycle
-- Executing [0320209...@inputinterior.se:4]
NoOp(SIP/0317998985-0050, ) in new stack
-- Executing [0320209...@inputinterior.se:5]
NoOp(SIP/0317998985-0050, ) in new stack
-- Executing [0320209...@inputinterior.se:6]
NoOp(SIP/0317998985-0050, Kinna) in new stack
-- Executing [0320209...@inputinterior.se:7]
Queue(SIP/0317998985-0050, Goteborg,rtT) in new stack
QUEUENAME is working the way i am excpecting but MEMBERINTERFACE and
MEMBERNAME is not, or am I wrong?
From: magnu...@inputinterior.se
Sent: Sunday, February 20, 2011 8:05 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions
Hi!
Did play around with queues and need some help. I thought that
MEMBERINTERFACE and MEMBERNAME should be set to the “device” the call was
queued to not the device that called the queue, or do i miss something?
Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on
2011-01-31 13:38:23 UTC
0317998985 calls Kinna (0320209030)
Tomas Ekman (SIP/0317998972) receives the call but don’t answer.
When the queue “timeout” I would like to get the name of the device that
didn’t answered, in my case: SIP/0317998972.
${MEMBERINTERFACE} gives me the name of the device that called in.
queue show Kinna
Kinna has 0 calls (max unlimited) in 'rrmemory' strategy (4s holdtime, 2s
talktime), W:0, C:1, A:13, SL:0.0% within 0s
Members:
Tomas Ekman (SIP/0317998972) with penalty 1 (dynamic) (Not in use) has
taken no calls yet
No Callers
queues.conf
---
[general]
;
autofill=yes
keepstats=yes
setinterfacevar=yes
;
[Kinna]
retry=5
ringinuse=no
strategy=rrmemory
timeout=20
wrapuptime=120
extensions.conf
---
exten = 0320209030,1,Answer()
exten =
0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT))
exten = 0320209030,n,Queue(Kinna,nrtT)
exten = 0320209030,n,NoOp(${MEMBERINTERFACE})
exten = 0320209030,n,NoOp(${MEMBERNAME})
exten = 0320209030,n,Queue(Goteborg,rtT)
exten = 0320209030,n,Hangup()
CLI
== Using SIP RTP CoS mark 5
-- Executing [0320209...@inputinterior.se:1]
Answer(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:2]
ExecIf(SIP/0317998985-0033, 0?Queue(Goteborg,rtT)) in new stack
-- Executing [0320209...@inputinterior.se:3]
Queue(SIP/0317998985-0033, Kinna,nrtT) in new stack
== Using SIP RTP CoS mark 5
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
-- Executing [0320209...@inputinterior.se:4]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:5]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:6]
Queue(SIP/0317998985-0033, Goteborg,rtT) in new stack
== Spawn extension (inputinterior.se, 0320209030, 6) exited non-zero on
'SIP/0317998985-0033'
Could any help me understand what I am doing wrong?
/Magnus
--
Re: [asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6
*Bump* No takers? Perhaps no-one else thinks this is a bug? Regards, Steve On 7 February 2011 16:45, Steve Davies davies...@gmail.com wrote: Hi, The following IAX config (slightly edited) causes an issue for me in version 1.6.2.16.1, where my CDR data is unreliable. [user1] type=friend auth=md5 accountcode=user1 notransfer=yes context=context1 host=10.0.0.250 username=user1 secret=secret1 disallow=all allow=alaw [user2] type=friend auth=md5 accountcode=user2 notransfer=yes context=context2 host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/24 username=user2 secret= disallow=all allow=alaw If a call comes in from 10.0.0.250, using username user2 and with no password, then it is correctly authenticated against the [user2] section. Accountcode is set to user2 Context is set to context2 and the call mostly proceeds correctly, BUT the source channel name is set to IAX2/user1-, which is then seen both in the dialplan debug output, and in the CDR. I would expect the channel name to reflect the section name that was used to authenticate the call ie. IAX2/user2-; I specifically put a password onto [user1] so there is no possibility that the call is authenticating there. Am I missing something? Or is this a bug? Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)
On 2/02/11 7:05 AM, Olivier wrote: Hi Matt, Too bad I can't be more helpful on this but could work around this issue ? Nah, in the end I just learnt how to use LCR with mISDN. I upgraded DAHDI, LibPRI, Asterisk to latest versions and still no go - although the errors stopped happening. I'm going to try again this weekend - with a different b410P card - even though it works with mISDN :) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber Issue
On 11/02/11 6:54 PM, William Stillwell wrote: I was getting unable to make channel.. We couldn't get it to work properly until we upgraded to Asterisk 1.8 at which stage it magically started working (with the same configs etc). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber Issue
I was also informed it only works in 1.8, I think there was a protocol change I think that wasn't back ported to 1.6. Also here: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google Calling using Google Voice or via the Google Talk web client requires the use of Asterisk 1.8.1.1 or greater. The 1.6.x versions of Asterisk only support calls made using the legacy GoogleTalk external client. --- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Sunday, February 20, 2011 5:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Gtalk/Jabber Issue On 11/02/11 6:54 PM, William Stillwell wrote: I was getting unable to make channel.. We couldn't get it to work properly until we upgraded to Asterisk 1.8 at which stage it magically started working (with the same configs etc). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber Issue
And confirmed, just upgraded to 1.8.x.x branch, outbound/inbound working fine. Now, no outbound callerid? Shows 'unknown caller' on called party's handsets. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Sunday, February 20, 2011 6:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Gtalk/Jabber Issue I was also informed it only works in 1.8, I think there was a protocol change I think that wasn't back ported to 1.6. Also here: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google Calling using Google Voice or via the Google Talk web client requires the use of Asterisk 1.8.1.1 or greater. The 1.6.x versions of Asterisk only support calls made using the legacy GoogleTalk external client. --- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Sunday, February 20, 2011 5:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Gtalk/Jabber Issue On 11/02/11 6:54 PM, William Stillwell wrote: I was getting unable to make channel.. We couldn't get it to work properly until we upgraded to Asterisk 1.8 at which stage it magically started working (with the same configs etc). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX on PRI to MFCR2
Thanks for the insight will try that one. Is there any suggested value for this parameter that best work well on fax? Regards, Mac From: Moises Silva moises.si...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 19, 2011 10:05:45 PM Subject: Re: [asterisk-users] FAX on PRI to MFCR2 On Fri, Feb 18, 2011 at 3:23 AM, leonimar cape leo_mac...@yahoo.com wrote: Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is connected. Fax passing thru this connection are not getting thru and always getting drop. FYI: all of the 8xE1s are currently up with two using mfcr2 and the rest is ISDN pri. My guess is that using 2 different cards from 2 different manufacturers for FAX (which is sensitive to time slips) is probably a bad idea because most likely is not possible to synchronize their clocks. If you were using 2 Sangoma ports you can sync the ports with the TE_REF_CLOCK parameter. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber Issue
Unknown Caller most likely refers to the CID Name, CID Number should be provided as your Google Voice number. On 2/20/2011 5:53 PM, William Stillwell wrote: And confirmed, just upgraded to 1.8.x.x branch, outbound/inbound working fine. Now, no outbound callerid? Shows 'unknown caller' on called party's handsets. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Sunday, February 20, 2011 6:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Gtalk/Jabber Issue I was also informed it only works in 1.8, I think there was a protocol change I think that wasn't back ported to 1.6. Also here: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google Calling using Google Voice or via the Google Talk web client requires the use of Asterisk 1.8.1.1 or greater. The 1.6.x versions of Asterisk only support calls made using the legacy GoogleTalk external client. --- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Sunday, February 20, 2011 5:39 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Gtalk/Jabber Issue On 11/02/11 6:54 PM, William Stillwell wrote: I was getting unable to make channel.. We couldn't get it to work properly until we upgraded to Asterisk 1.8 at which stage it magically started working (with the same configs etc). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gtalk/Jabber Issue
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Sunday, February 20, 2011 10:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Gtalk/Jabber Issue
Unknown Caller most likely refers to the CID Name, CID Number should
be provided as your Google Voice number.
I ended up doing the following:
Outbound rule:
Exten =
_NX,1,Dial(gtalk/value_in_jabber.conf/+1(myGoogleVoice#)@voice.googl
e.com,30,D(ww2www${EXTEN}#ww))
This will call the gv service, and then dial out that way, and the remote
receiving party will see my gv #
And on inbound to bypass call screening question:
exten = s,1,NoOp(The Current Time is..:
${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,n,NoOp(The CallerID(num) is.: ${CALLERID(NUM)})
exten = s,n,NoOp(The CallerID(name) is: ${CALLERID(NAME)})
exten = s,n,Answer()
exten = s,n,Wait(5)
exten = s,n,SendDTMF(1)
exten = s,n,Dial(...
--
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Re: [asterisk-users] Gtalk/Jabber Issue
William,
It still looks like something is not properly set with your account on
Google Voice. Have you had a chance to follow the recommendations I
gave you earlier in the thread?
If the account is properly set the dial string will need to look like
this, gtalk/jabber-conf-section-name/+$OUTNUM$@voice.google.com
where $OUTNUM$ is a called number in the international format.
On the receiving end the call will come with an empty CID Number, but
with the CID Name which looks like this:
+1551...@voice.google.com/srvres-MTAuMjE4LjIuMTk3Ojk4MzM=
Just cut all prior to @ as a CID Number. See
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
Also you do not need to wait 5 seconds. 1 or 2 is sufficient.
-Vladimir
On 2/20/2011 10:40 PM, William Stillwell wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Sunday, February 20, 2011 10:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Gtalk/Jabber Issue
Unknown Caller most likely refers to the CID Name, CID Number should
be provided as your Google Voice number.
I ended up doing the following:
Outbound rule:
Exten =
_NX,1,Dial(gtalk/value_in_jabber.conf/+1(myGoogleVoice#)@voice.googl
e.com,30,D(ww2www${EXTEN}#ww))
This will call the gv service, and then dial out that way, and the remote
receiving party will see my gv #
And on inbound to bypass call screening question:
exten = s,1,NoOp(The Current Time is..:
${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,n,NoOp(The CallerID(num) is.: ${CALLERID(NUM)})
exten = s,n,NoOp(The CallerID(name) is: ${CALLERID(NAME)})
exten = s,n,Answer()
exten = s,n,Wait(5)
exten = s,n,SendDTMF(1)
exten = s,n,Dial(...
--
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