[asterisk-users] asterisk 1.8 on Solaris/sparc
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. The system itself is happy and phone calls (between two parties) seem fine. Unfortunately, when a caller listens to a Playback recording, there seems to be moments of stutter - perhaps 1 second of stutter for every 10 seconds of Playback. The stutter is not consistent at the same point of the playback file. To eliminate encoding as an issue, I have only codec_ulaw/format_pcm loaded and the recording is ulaw. I've niced down the asterisk process to -19 even though I don't see asterisk taking more than 3% cpu. Is this behavior indicative of a timing problem? loading res_timing_pthread.so makes things horribly worse. i don't believe any other software timer is available for Solaris/sparc, right ? other thoughts ? Thanks, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Inconsistency in CDR between NO ANSWER and BUSY calls
On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.7 My dialplan for an inbound number is along the lines of [default] exten = idenfier,1,Goto(specific-context,s,1) [specific-context] exten = s,1,NoOp() exten = s,2,Dial(SIP/some-extenion,20) I have been testing the following 2 scenarios: 1) Caller calls in to identifier, caller hangs up (NO ANSWER) 2) Caller calls in to identifier, callee rejects (BUSY) In both scenarios the dialplan works properly and dials 'some-extension'. However, there is some divergence with what is entered into the CDR. In both scenarios the following are the same (as they should be) a) lastapp b) lastdata But, in scenario 1 the dcontext is 'specific-context' (this is what I would expect) and in scenario 2 the dcontext remains 'default' even though the call moved to a different context. This cannot possibly be intentional and it is causing problems with our set up. Has anyone else experienced this? Is it actually correct behaviour and if so, why? If it is a bug, has it already been raised? Thanks in advance Would I be better off asking this question of the dev community? Thanks Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Asterisk with OpenBTS and mobile phone
Hey Ioan,
thanks for your answer.
It helped a little bit but I have no idea what exactly could work wrong.
My new situation:
*CLI originate SIP/123456789101112 application MusicOnHold
== Using SIP RTP CoS mark 5
-- Got SIP response 482 Loop Detected back from 192.168.0.102:5060
[Jul 18 10:38:27] WARNING[4615]: chan_sip.c:3873 __sip_autodestruct:
Autodestruct on dialog '
446588d34c8b0e2d1920fec416ef0b5d@192.168.0.102:5060' with owner in place
(Method: INVITE)
*CLI sip show peers
Name/username HostDyn
Forcerport ACL Port Status
123456789101112/6202 192.168.0.102
N 5060 OK (1 ms)
6000/6000 192.168.0.102D
N 5061 Unmonitored
6001/6001 192.168.0.102D
N 5061 Unmonitored
*CLI sip show channels
Peer User/ANR Call ID Format
Hold Last MessageExpiry Peer
192.168.0.102(None) 2dab9ef669bc9a4 0x0 (nothing)
No Rx: OPTIONSguest
1 active SIP dialog
I thought with 6201 I could build a connection to Asterisk. In the
extensions.conf and in the Asterisk-GUI the numbers from 6000 - 6300 (not
all, just a frew of them) are shown so I choosed one of them like I did
with the softphones.
asterisk -rx doesn't work.
What do you think is wrong with my extensions.conf?
Best regards.
Ellen
On Fri, Jul 13, 2012 at 4:06 PM, Ioan Indreias indre...@gmail.com wrote:
On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar
ellen.apolinar...@googlemail.com wrote:
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile
phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
OpenBTS works and I can send sms from the OpenBTS server to the
mobile phone. What I also need is a call between Asterisk and OpenBTS.
I have also two soft phones which works with Asterisk. And also OpenBSC
is working with Asterisk successfully (OpenBSC is another project).
Perhaps you can help me because I think it is an issue with Asterisk.
sip.conf:
;SIP-Phones (Twinkle)
[user1]
callerid = 6000
username = 6000
secret = 6000
canreinvite = no
type = friend
context = phones
allow = all
host = dynamic
dtmfmode = info
[user2]
callerid = 6001
username = 6001
secret = 6001
canreinvite = no
type = friend
context = phones
allow = all
host = dynamic
dtmfmode = info
; Mobile phone
[123456789101112]
callerid = 6201
username = 6201
secret = 6201
canreinvite = no
type = friend
context = sip_external
;context = open-bts
disallow = all
allow = gsm
host = 192.168.0.102
domain = 192.168.0.102
dtmfmode = info
extensions.conf
[internal]
exten = s,1,Verbose(1|Echo test application)
exten = s,n,Echo()
exten = s,n,Hangup()
exten = 6000,1,Verbose(1|Extension 6000)
exten = 6000,n,Dial(SIP/user1,30)
exten = 6000,n,Hangup()
exten = 6001,1,Verbose(1|Extension 6001)
exten = 6001,n,Dial(SIP/user2,30)
exten = 6001,n,Hangup()
[phones]
include = internal
include = default
[open-bts]
exten = 6002,1,Playback(demo-echotest)
exten = 6002,n,Echo
exten = 6002,n,Playback(demo-echodone)
exten = 6002,n,HangUp
[sip_external]
exten = 6201,1,Macro(dialGSM,123456789101112)
[macro-dialGSM]
exten = s,1,Dial(SIP/${ARG1},20)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-CONGESTION,1,Congestion (30)
exten = s-CHANUNAVAIL,1,Read(extension_digits,pbx-invalid)
exten = s-CHANUNAVAIL,n,GoTo(open-bts,${extension_digits},1)
I have tried both contexts, [open-bts] and [sip_external] and both don't
work
If I want to call the mobile phone (6201) with a Twinkle soft phone
(6000)
I get following message in the CLI-window from Asterisk:
== Using SIP RTP CoS mark 5
-- Executing [6201@DLPN_DialPlan1:1] Macro(SIP/6000-0013,
stdexten,6201,SIP/6201) in new stack
-- Executing [s@macro-stdexten:1] Set(SIP/6000-0013,
__DYNAMIC_FEATURES=) in new stack
[Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected '=', expecting
$end;
Input:
= 1
^
[Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyerror: If
you
have questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@macro-stdexten:2] GotoIf(SIP/6000-0013,
?5:3) in new stack
-- Goto (macro-stdexten,s,3)
-- Executing [s@macro-stdexten:3] Dial(SIP/6000-0013,
SIP/6201,20,) in new stack
[Jul 12 12:14:29] WARNING[7092]: app_dial.c:2274 dial_exec_full:
Unable to create channel of type 'SIP' (cause
[asterisk-users] Telecom HU cannot callforward to external number
Hi List! I have a Problem with Telecom Hungary, if I set a callforwarding on the Snom, to an external number (mobile). Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30 When I call the Snom (Extension 68), it responds with 302 moved temporarily, and Asterisk try to dial out over the LOCAL channel using DAHDI. I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup request, cause 21 Here is cli output: -- Accepting call from 'callerid' to '68' on channel 0/1, span 1 -- Executing [s@macro-station-fallback-Q-VM:5] Dial(DAHDI/1-1, SIP/68|15|tTW) in new stack -- Called 68 -- Got SIP response 302 Moved Temporarily back from 10.70.x.xxx -- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr@snom68' (thanks to SIP/68-76b8) -- Executing [*1mobilenr@snom68:1] Macro(Local/*1mobilenr@snom68-2fe3,2, dialout-dahdi-test|mobilenr|g1|) in new stack -- Executing [s@macro-dialout-dahdi-test:1] Set(Local/*1mobilenr@snom68-2fe3,2, CALLERID(number)=) in new stack -- Executing [s@macro-dialout-dahdi-test:2] Dial(Local/*1mobilenr@snom68-2fe3,2, DAHDI/g1/mobilenr||) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/mobilnr -- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr@snom68-2fe3,2 -- Local/*1mobilenr@snom68-2fe3,1 is proceeding passing it to DAHDI/1-1 -- Channel 0/2, span 1 got hangup request, cause 21 -- DAHDI/2-1 is circuit-busy -- Hungup 'DAHDI/2-1' == Everyone is busy/congested at this time (1:0/1/0) I have also an output from pri intense debug - But I think the Telecom is just not accepting the outgoing call. What do you think? thanks yours christian -- _ -- 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] Telecom HU cannot callforward to external number
Mebe your operator doesnt like the CallerID(num) set as NULL just remove the callerid(num) statement and let the standard callerId get set by network. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Wed, Jul 18, 2012 at 4:23 PM, Christian Gansberger christian.gansber...@accm.at wrote: Hi List! I have a Problem with Telecom Hungary, if I set a callforwarding on the Snom, to an external number (mobile). Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30 When I call the Snom (Extension 68), it responds with 302 moved temporarily, and Asterisk try to dial out over the LOCAL channel using DAHDI. I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup request, cause 21 Here is cli output: -- Accepting call from 'callerid' to '68' on channel 0/1, span 1 -- Executing [s@macro-station-fallback-Q-VM:5] Dial(DAHDI/1-1, SIP/68|15|tTW) in new stack -- Called 68 -- Got SIP response 302 Moved Temporarily back from 10.70.x.xxx -- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr@snom68' (thanks to SIP/68-76b8) -- Executing [*1mobilenr@snom68:1] Macro(Local/*1mobilenr@snom68-2fe3,2, dialout-dahdi-test|mobilenr|g1|) in new stack -- Executing [s@macro-dialout-dahdi-test:1] Set(Local/*1mobilenr@snom68-2fe3,2, CALLERID(number)=) in new stack -- Executing [s@macro-dialout-dahdi-test:2] Dial(Local/*1mobilenr@snom68-2fe3,2, DAHDI/g1/mobilenr||) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/mobilnr -- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr@snom68-2fe3 ,2 -- Local/*1mobilenr@snom68-2fe3,1 is proceeding passing it to DAHDI/1-1 -- Channel 0/2, span 1 got hangup request, cause 21 -- DAHDI/2-1 is circuit-busy -- Hungup 'DAHDI/2-1' == Everyone is busy/congested at this time (1:0/1/0) I have also an output from pri intense debug - But I think the Telecom is just not accepting the outgoing call. What do you think? thanks yours christian -- _ -- 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
[asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten = _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)
I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:
1. Asterisk does NOT send a REINVITE with the t38 offered. Reading the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.
2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.
Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.
I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.
Does anybody have any experience in making this work?
Thank you!
Alex
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Telecom HU cannot callforward to external number
Unfortunately not, I already tried different forms callerid(num). Always the same error. I came across this entry in asterisk changelogs - maybe an update of asterisk will help. * Asterisk 1.4.36-rc1 Released. 2010-08-20 16:46 + [r283048-283123] Richard Mudgett rmudg...@digium.com * channels/chan_dahdi.c: Merged revision 278274 from https://origsvn.digium.com/svn/asterisk/trunk .. r278274 | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 line Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR. .. * channels/chan_dahdi.c: Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8) (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) thx christian On 18 July 2012 13:07, Mitul Limbani mi...@enterux.in wrote: Mebe your operator doesnt like the CallerID(num) set as NULL just remove the callerid(num) statement and let the standard callerId get set by network. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Wed, Jul 18, 2012 at 4:23 PM, Christian Gansberger christian.gansber...@accm.at wrote: Hi List! I have a Problem with Telecom Hungary, if I set a callforwarding on the Snom, to an external number (mobile). Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30 When I call the Snom (Extension 68), it responds with 302 moved temporarily, and Asterisk try to dial out over the LOCAL channel using DAHDI. I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup request, cause 21 Here is cli output: -- Accepting call from 'callerid' to '68' on channel 0/1, span 1 -- Executing [s@macro-station-fallback-Q-VM:5] Dial(DAHDI/1-1, SIP/68|15|tTW) in new stack -- Called 68 -- Got SIP response 302 Moved Temporarily back from 10.70.x.xxx -- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr@snom68' (thanks to SIP/68-76b8) -- Executing [*1mobilenr@snom68:1] Macro(Local/*1mobilenr@snom68-2fe3,2, dialout-dahdi-test|mobilenr|g1|) in new stack -- Executing [s@macro-dialout-dahdi-test:1] Set(Local/*1mobilenr@snom68-2fe3,2, CALLERID(number)=) in new stack -- Executing [s@macro-dialout-dahdi-test:2] Dial(Local/*1mobilenr@snom68-2fe3,2, DAHDI/g1/mobilenr||) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/mobilnr -- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr@snom68-2fe3 ,2 -- Local/*1mobilenr@snom68-2fe3,1 is proceeding passing it to DAHDI/1-1 -- Channel 0/2, span 1 got hangup request, cause 21 -- DAHDI/2-1 is circuit-busy -- Hungup 'DAHDI/2-1' == Everyone is busy/congested at this time (1:0/1/0) I have also an output from pri intense debug - But I think the Telecom is just not accepting the outgoing call. What do you think? thanks yours christian -- _ -- 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] Using Asterisk 10.6 as a T38 Fax gateway
I forgot to ask: Do I have to load res_fax or app_fax to use the T38 gateway capability? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to work around asterisk ss7
Hello, Can someone give me an understanding about E1 with ISUP on CCS 7 signalling? Is it possible with asterisk + digium card and how Regards, Ashish -- _ -- 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] How to work around asterisk ss7
you need to either use chan_ss7 or libss7. Also look for mailing list archives of asterisk-ss7 Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal ashisha...@gmail.comwrote: Hello, Can someone give me an understanding about E1 with ISUP on CCS 7 signalling? Is it possible with asterisk + digium card and how Regards, Ashish -- _ -- 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] How to work around asterisk ss7
You can use asterisk 1.6+ and libss7 for this functionality. Any Digium or Sangoma card working ok on this setup. Currently i am using it on both of them. On Wed, Jul 18, 2012 at 5:14 PM, Ashish Agarwal ashisha...@gmail.com wrote: Hello, Can someone give me an understanding about E1 with ISUP on CCS 7 signalling? Is it possible with asterisk + digium card and how Regards, Ashish -- _ -- 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 -- Bharat Lalcheta -- _ -- 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] Using Asterisk 10.6 as a T38 Fax gateway
- Original Message -
From: Alejandro Recarey a...@recarey.org
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten = _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)
The correct setting is not FAXOPT(t38gateway) - that is not a valid parameter
to pass to the FAXOPT function. As you mention below, the correct setting
is Set(FAXOPT(gateway)=yes). The optional timeout is fine.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_FAXOPT
I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:
1. Asterisk does NOT send a REINVITE with the t38 offered. Reading
the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.
Have you confirmed that Asterisk does not send the re-INVITE using either
a packet sniffer or by monitoring the log with 'sip set debug on'? Without
seeing the SIP message traffic and a DEBUG log, its hard to say what
might be the cause of your issues.
Typically, I would expect to see something like the following in a DEBUG log:
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: detected v21 preamble from
SIP/ast1-g711-0001
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: requesting T.38 for gateway session
for SIP/ast1-t38-
Note that this also answers your question in a subsequent e-mail: you
should be using res_fax, with either res_fax_spandsp or Fax for Asterisk.
2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.
Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.
I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.
There typically isn't a lot of configuration that is needed for T.38
gateway support. The necessary dialplan configuration is documented
here:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
Does anybody have any experience in making this work?
Thank you!
Alex
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls
- Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 18, 2012 3:13:13 AM Subject: Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.7 My dialplan for an inbound number is along the lines of [default] exten = idenfier,1,Goto(specific-context,s,1) [specific-context] exten = s,1,NoOp() exten = s,2,Dial(SIP/some-extenion,20) snip So, I mocked up what is essentially the same scenario using the following dialplan: [default] exten = 100,1,NoOp() same = n,Goto(new_context,s,1) [new_context] exten = s,1,NoOp() same = n,Dial(SIP/phone_b,10) Assume that I have two SIP peers, phone_a and phone_b, where phone_a dials extension 100 in context default. In my case, I had phone_a first call phone_b, and did not answer with phone_b. That resulted in a disposition of NOANSWER. I then had phone_a call phone_b and had phone_b explicitly reject phone_a's call; that resulted in a disposition of BUSY. In that case, with the csv and csv-custom CDR backends, I get the following CDR records: [csv] ,phone_a,100,default,phone_a phone_a,SIP/phone_a-,SIP/phone_b-0001,Dial,SIP/phone_b,10,2012-07-18 13:52:50,,2012-07-18 13:53:00,10,0,NO ANSWER,DOCUMENTATION,1342619570.0, ,phone_a,100,default,phone_a phone_a,SIP/phone_a-0002,SIP/phone_b-0003,Dial,SIP/phone_b,10,2012-07-18 13:53:10,,2012-07-18 13:53:16,6,0,BUSY,DOCUMENTATION,1342619590.2, [csv-custom] phone_a phone_a,phone_a,100,default,SIP/phone_a-,SIP/phone_b-0001,Dial,SIP/phone_b,10,2012-07-18 08:52:50,,2012-07-18 08:53:00,10,0,NO ANSWER,DOCUMENTATION,,1342619570.0,,0 phone_a phone_a,phone_a,100,default,SIP/phone_a-0002,SIP/phone_b-0003,Dial,SIP/phone_b,10,2012-07-18 08:53:10,,2012-07-18 08:53:16,6,0,BUSY,DOCUMENTATION,,1342619590.2,,2 You'll note that in both, the destination context is default. So at the very least, the records are consistent. I ran this test using the latest from 1.8 (1.8.15.0-rc1) - since the version you're using is older and 1.8 has had some bug fixes with respect to CDRs since then, that might explain the discrepancy. Why is destination context default and not new_context? The destination context is initially set when the CDR for the call is initialized - in which case, its initial value is default, since that's where the call entered. When a CDR is updated, the destination context is also updated to the channel's current context (or macrocontext). As it is, a CDR update is not the same thing as the CDR being ended, and does not always occur before the CDR is ended. In the case you've outlined, an explicit CDR update does not occur before the CDR record is ended - which results in the original context of default being recorded as opposed to the current context of the channel, new_context. snip Has anyone else experienced this? Is it actually correct behaviour and if so, why? If it is a bug, has it already been raised? A quick search through JIRA would answer this question. I don't believe anyone has filed a bug related to this issue. As far as it being the correct behavior - the behavior of the destination context feels more implied then well defined. Currently, the destination context is updated when the source and destination channels are bridged, as opposed to when the Dial is executed to the destination channel. If the bridge never occurs, then the destination context is the original context of the source channel, since that's the most information that is known at the time of CDR creation. While that behavior may not be what you desired, it is at least the current implementation. Thanks in advance Would I be better off asking this question of the dev community? Nope, as this isn't an Asterisk development question. Thanks Ish -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
On 07/18/2012 09:43 PM, Matthew Jordan wrote:
- Original Message -
From: Alejandro Recarey a...@recarey.org
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten = _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)
The correct setting is not FAXOPT(t38gateway) - that is not a valid parameter
to pass to the FAXOPT function. As you mention below, the correct setting
is Set(FAXOPT(gateway)=yes). The optional timeout is fine.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_FAXOPT
I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:
1. Asterisk does NOT send a REINVITE with the t38 offered. Reading
the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.
Have you confirmed that Asterisk does not send the re-INVITE using either
a packet sniffer or by monitoring the log with 'sip set debug on'? Without
seeing the SIP message traffic and a DEBUG log, its hard to say what
might be the cause of your issues.
Typically, I would expect to see something like the following in a DEBUG log:
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: detected v21 preamble from
SIP/ast1-g711-0001
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: requesting T.38 for gateway session
for SIP/ast1-t38-
Note that this also answers your question in a subsequent e-mail: you
should be using res_fax, with either res_fax_spandsp or Fax for Asterisk.
2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.
Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.
I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.
There typically isn't a lot of configuration that is needed for T.38
gateway support. The necessary dialplan configuration is documented
here:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
One thing that page doesn't mention is only spandsp supports T.38
gateway right now. The Digium FAX module does not.
Regards,
Steve
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Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls
On Wed, 2012-07-18 at 09:16 -0500, Matthew Jordan wrote: - Original Message - From: Ishfaq Malik i...@pack-net.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 18, 2012 3:13:13 AM Subject: Re: [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls On Wed, 2012-07-11 at 15:08 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.7 My dialplan for an inbound number is along the lines of [default] exten = idenfier,1,Goto(specific-context,s,1) [specific-context] exten = s,1,NoOp() exten = s,2,Dial(SIP/some-extenion,20) snip So, I mocked up what is essentially the same scenario using the following dialplan: [default] exten = 100,1,NoOp() same = n,Goto(new_context,s,1) [new_context] exten = s,1,NoOp() same = n,Dial(SIP/phone_b,10) Assume that I have two SIP peers, phone_a and phone_b, where phone_a dials extension 100 in context default. In my case, I had phone_a first call phone_b, and did not answer with phone_b. That resulted in a disposition of NOANSWER. I then had phone_a call phone_b and had phone_b explicitly reject phone_a's call; that resulted in a disposition of BUSY. In that case, with the csv and csv-custom CDR backends, I get the following CDR records: [csv] ,phone_a,100,default,phone_a phone_a,SIP/phone_a-,SIP/phone_b-0001,Dial,SIP/phone_b,10,2012-07-18 13:52:50,,2012-07-18 13:53:00,10,0,NO ANSWER,DOCUMENTATION,1342619570.0, ,phone_a,100,default,phone_a phone_a,SIP/phone_a-0002,SIP/phone_b-0003,Dial,SIP/phone_b,10,2012-07-18 13:53:10,,2012-07-18 13:53:16,6,0,BUSY,DOCUMENTATION,1342619590.2, [csv-custom] phone_a phone_a,phone_a,100,default,SIP/phone_a-,SIP/phone_b-0001,Dial,SIP/phone_b,10,2012-07-18 08:52:50,,2012-07-18 08:53:00,10,0,NO ANSWER,DOCUMENTATION,,1342619570.0,,0 phone_a phone_a,phone_a,100,default,SIP/phone_a-0002,SIP/phone_b-0003,Dial,SIP/phone_b,10,2012-07-18 08:53:10,,2012-07-18 08:53:16,6,0,BUSY,DOCUMENTATION,,1342619590.2,,2 You'll note that in both, the destination context is default. So at the very least, the records are consistent. I ran this test using the latest from 1.8 (1.8.15.0-rc1) - since the version you're using is older and 1.8 has had some bug fixes with respect to CDRs since then, that might explain the discrepancy. Why is destination context default and not new_context? The destination context is initially set when the CDR for the call is initialized - in which case, its initial value is default, since that's where the call entered. When a CDR is updated, the destination context is also updated to the channel's current context (or macrocontext). As it is, a CDR update is not the same thing as the CDR being ended, and does not always occur before the CDR is ended. In the case you've outlined, an explicit CDR update does not occur before the CDR record is ended - which results in the original context of default being recorded as opposed to the current context of the channel, new_context. That makes sense to me snip Has anyone else experienced this? Is it actually correct behaviour and if so, why? If it is a bug, has it already been raised? A quick search through JIRA would answer this question. I don't believe anyone has filed a bug related to this issue. As far as it being the correct behavior - the behavior of the destination context feels more implied then well defined. Currently, the destination context is updated when the source and destination channels are bridged, as opposed to when the Dial is executed to the destination channel. If the bridge never occurs, then the destination context is the original context of the source channel, since that's the most information that is known at the time of CDR creation. While that behavior may not be what you desired, it is at least the current implementation. Fair enough, now that I know why it behaves as it does, I can go about making it behave the way I want it to. Your answers have been very helpful and I really appreciate it. Thanks Ish -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w:
Re: [asterisk-users] Call recording and transfer issue (asterisk 1.8)
On Thu, 2012-04-19 at 12:20 +0100, Ishfaq Malik wrote: Hi I'm having a problem with the entirety of a call being recorded in the following scenario I'm using asterisk 1.8.7.0 Person A (asterisk peer) calls Person B (not on asterisk, real world number via a SIP trunk) Mixmonitor is invoked by Person A in the outbound context and AUDIOHOOK_INHERIT(MixMonitor)=yes is also set Person a transfers Person B to Person C (another asterisk peer) Person A is no longer involved in the call and the call is bridged between Person B and Person C The call recording stops as soon as Person A hangs up, even though AUDIOHOOK_INHERIT is set Is there any way we can get the entire call recorded in one file? Thanks in advance Ish Has anyone else encountered this as it's becoming a real problem. Does anyone know a way of getting continuity of call recording in this scenario? -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Using Asterisk 10.6 as a T38 Fax gateway
On 07/18/2012 06:30 AM, Alejandro Recarey wrote:
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten = _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)
I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:
You say they don't work, but you don't provide any details (console
output, log messages, etc.) The configuration you have provided above is
*required* for T.38 support and T.38 gateway mode. If it's not working,
we are going to need more details about what is actually happening (if
anything is at all).
1. Asterisk does NOT send a REINVITE with the t38 offered. Reading the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.
This is expected behavior. Proper implementations of T.38 require that
the gateway in front of the *called* endpoint monitor for FAX tones and
initiate the switch to T.38 mode. In your configuration, that would be
your carrier's gateway, assuming it is terminating the call to a
non-T.38 endpoint. If your carrier is handing off the call to another
SIP provider, then the responsibility lies with them, and so on.
However, Asterisk's T.38 gateway functionality should still detect the
V.21 preamble generated by the called FAX endpoint and initiate a switch
to T.38, if the carrier does not do it first. If this is not happening,
we'll need to see logs and console output to figure out why. What codec
are you using for your SIP calls?
2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.
This is bizarre; there is no specification anywhere that would indicate
that a carrier should do this, and there are plenty of documents
describing how it is a *bad* idea to offer a second media stream for
T.38 in the initial INVITE of a call. I would urge you to ask them to
reconsider this behavior.
Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.
This is not as reliable as the far-end gateway doing it, especially if
the codec in use for the VoIP leg(s) of the call distorts the V.21
preamble in any significant way.
I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.
The documentation on the wiki is short, but it's complete. Enabling T.38
gateway functionality in Asterisk 10 is in fact pretty simple :-)
Problems arise, as they always do in T.38-land, because no two T.38
implementations are the same, and the choices made by carriers,
gateway/softswitch/SBC manufacturers, and others, result in
interoperability problems.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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[asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored. Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid modem. -- _ -- 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] Asterisk 1.8.13 / res_fax / res_fax_digium
On 07/18/2012 10:06 AM, Eric Wieling wrote: We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored. Is v34 only supported with SpanDSP? Those docs are in error. V.34 is not supported. I'll notify our documentation people. Thanks for the report. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium
Thank you. While you are at it, ask them to document where the audio / data from fax set g711cap| t38cap on is saved to. 8-) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, July 18, 2012 11:28 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium On 07/18/2012 10:06 AM, Eric Wieling wrote: We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored. Is v34 only supported with SpanDSP? Those docs are in error. V.34 is not supported. I'll notify our documentation people. Thanks for the report. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote party ID - sort of working...
Hi, I'm trying to set my system to set a caller id using the diaplan when calling an internal extension. In other words, when I dial Joe Smith's extension I want my own phone to show Joe Smith 555. I have sort of managed that in the sense that my phone shows Joe Smith's caller id based on his sip.conf callerid. But I need this to be done programmatically through the dial plan (Let's say I want to show Joe Smith or just Joe based on some condition) I have this in the relevant dialplan snippet: exten = 123,1,Verbose(1,Test) exten = 123,n,Set(CONNECTEDLINE(number,i)=555-555-) exten = 123,n,Set(rclidname=TestingB 123-444-) exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Set(CONNECTEDLINE(name,i)=Testing) exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Dial(SIP/joesmithpolycomphone,20) exten = 123,n,Hangup() I am always seeing remotepolycomphone's callerid number and name as entered in sip.conf, not Testing 555-555-, neither am I seeing TestingB 123-444-. What am I missing for it for accept my dialplan remote-id name and number? Regards, Mike -- _ -- 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] Remote party ID - sort of working...
Remove the ,i to start with. Do you have the various rpid related options in sip.conf set? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Wednesday, July 18, 2012 12:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Remote party ID - sort of working... Hi, I'm trying to set my system to set a caller id using the diaplan when calling an internal extension. In other words, when I dial Joe Smith's extension I want my own phone to show Joe Smith 555. I have sort of managed that in the sense that my phone shows Joe Smith's caller id based on his sip.conf callerid. But I need this to be done programmatically through the dial plan (Let's say I want to show Joe Smith or just Joe based on some condition) I have this in the relevant dialplan snippet: exten = 123,1,Verbose(1,Test) exten = 123,n,Set(CONNECTEDLINE(number,i)=555-555-) exten = 123,n,Set(rclidname=TestingB 123-444-) exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Set(CONNECTEDLINE(name,i)=Testing) exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Dial(SIP/joesmithpolycomphone,20) exten = 123,n,Hangup() I am always seeing remotepolycomphone's callerid number and name as entered in sip.conf, not Testing 555-555-, neither am I seeing TestingB 123-444-. What am I missing for it for accept my dialplan remote-id name and number? Regards, Mike -- _ -- 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] Remote party ID - sort of working...
I’m trying to set my system to set a caller id using the diaplan when calling an internal extension. In other words, when I dial Joe Smith’s extension I want my own phone to show “Joe Smith 555”. I have sort of managed that in the sense that my phone shows Joe Smith’s caller id based on his sip.conf callerid. But I need this to be done programmatically through the dial plan (Let’s say I want to show “Joe Smith” or just “Joe” based on some condition) I have this in the relevant dialplan snippet: exten = 123,1,Verbose(1,Test) exten = 123,n,Set(CONNECTEDLINE(number,i)=555-555-) exten = 123,n,Set(rclidname=TestingB 123-444-) This line is just setting an ordinary channel variable. What do you think is supposed to use this value? exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Set(CONNECTEDLINE(name,i)=Testing) Please read about usage of the ,i in [1]. If anything you should have the ,i on *all* of the CONNECTEDLINE lines *except* the last one before a Dial. exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Dial(SIP/joesmithpolycomphone,20) Since you are not using the 'I' option on Dial here, your preset CONNECTEDLINE information is being overwritten by what is sent by SIP/joesmithpolycomphone when it answers. exten = 123,n,Hangup() I am always seeing remotepolycomphone’s callerid number and name as entered in sip.conf, not “Testing 555-555-”, neither am I seeing “TestingB 123-444-”. What am I missing for it for accept my dialplan remote-id name and number? [1] https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Richard -- _ -- 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] Asterisk 1.8.13 / res_fax / res_fax_digium
On 07/18/2012 10:51 AM, Eric Wieling wrote: Thank you. While you are at it, ask them to document where the audio / data from fax set g711cap| t38cap on is saved to. 8-) That is documented in the CLI help for the commands themselves; the capture files are placed into subdirectories of the main Asterisk log directory. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote party ID - sort of working...
I m trying to set my system to set a caller id using the diaplan when calling an internal extension. In other words, when I dial Joe Smith s extension I want my own phone to show Joe Smith 555 . I have sort of managed that in the sense that my phone shows Joe Smith s caller id based on his sip.conf callerid. But I need this to be done programmatically through the dial plan (Let s say I want to show Joe Smith or just Joe based on some condition) [1] https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Informa tion Richard, For future ref: I used the straight forward dial-through example, pretty much as is on the link you included (thank you!), but although the number worked fine the name did not. I had to use exten = 124,n,Set(CONNECTEDLINE(all,i)=Name 555-555-) instead of a separate name and number priority. ...so the wiki might be wrong (or there might be a bug in my 1.8.x version). But it works now, thanks to you for pointing me in the right direction, Mike -- _ -- 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] Remote party ID - sort of working...
exten = 124,n,Set(CONNECTEDLINE(all,i)=Name 555-555-) instead of a
separate name and number priority.
An example of my line is:
Set(CONNECTEDLINE(all)=${cid.name} ${ARG1})
Doug
--
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Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Remote party ID - sort of working...
Why would you NOT want the connectedline info sent immediately? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, July 18, 2012 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote party ID - sort of working... I’m trying to set my system to set a caller id using the diaplan when calling an internal extension. In other words, when I dial Joe Smith’s extension I want my own phone to show “Joe Smith 555”. I have sort of managed that in the sense that my phone shows Joe Smith’s caller id based on his sip.conf callerid. But I need this to be done programmatically through the dial plan (Let’s say I want to show “Joe Smith” or just “Joe” based on some condition) I have this in the relevant dialplan snippet: exten = 123,1,Verbose(1,Test) exten = 123,n,Set(CONNECTEDLINE(number,i)=555-555-) exten = 123,n,Set(rclidname=TestingB 123-444-) This line is just setting an ordinary channel variable. What do you think is supposed to use this value? exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Set(CONNECTEDLINE(name,i)=Testing) Please read about usage of the ,i in [1]. If anything you should have the ,i on *all* of the CONNECTEDLINE lines *except* the last one before a Dial. exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Dial(SIP/joesmithpolycomphone,20) Since you are not using the 'I' option on Dial here, your preset CONNECTEDLINE information is being overwritten by what is sent by SIP/joesmithpolycomphone when it answers. exten = 123,n,Hangup() I am always seeing remotepolycomphone’s callerid number and name as entered in sip.conf, not “Testing 555-555-”, neither am I seeing “TestingB 123-444-”. What am I missing for it for accept my dialplan remote-id name and number? [1] https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Richard -- _ -- 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] asterisk 1.8 on Solaris/sparc
On 7/18/2012 2:27 AM, Jeremy Kister wrote: I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10. .. ok, if the system weren't Solaris - let's say it was Debian Linux, what would be on the list of things to check for ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Remote party ID - sort of working...
exten = 123,1,Verbose(1,Test) exten = 123,n,Set(CONNECTEDLINE(number,i)=555-555-) exten = 123,n,Set(rclidname=TestingB 123-444-) This line is just setting an ordinary channel variable. What do you think is supposed to use this value? exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Set(CONNECTEDLINE(name,i)=Testing) Please read about usage of the ,i in [1]. If anything you should have the ,i on *all* of the CONNECTEDLINE lines *except* the last one before a Dial. exten = 123,n,Set(CONNECTEDLINE(pres)=allowed) exten = 123,n,Dial(SIP/joesmithpolycomphone,20) Since you are not using the 'I' option on Dial here, your preset CONNECTEDLINE information is being overwritten by what is sent by SIP/joesmithpolycomphone when it answers. exten = 123,n,Hangup() [1] https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information Why would you NOT want the connectedline info sent immediately? So you don't have the channel driver's protocol needlessly generating update traffic on partial information. Have the protocol update the whole party ID information at once instead of piecemeal. Depending upon the channel protocol involved, connected line updates look like call transfers to the peer. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can not get my Eicon Diva running with Asterisk...
Hi Guys, asterisk drive me crazy! Now I have tried to use FreePBX but it require MySQL which I can not install du to a conflict with PostgreSQL. Does someone know, how to configure FreePBX to use PostgreSQL? Or does someone know another Asterisk Web-Frontend, without Database? It is realy not funny, to force users to install this monster on an ARM Microcontroller. I need only enterprise internal stuff to 1) access my 4 Vodafone EasyBox 803A using ISDN and the Eicon Diva 4port v2 Server Card 2) access a 1port HFC Card to connect some ISDN Telephones 3) access my account (20 Telephone numbers) on http://www.sipgate.de/ 4) let me use my CISCO CP-3905 SIP phones on the LAN 5) access the VoIP server of my ISP Alice/Hansenet I have no customers which must accounted or such... Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux Internet Service Provider, Cloud Computing http://www.itsystems.tamay-dogan.net/ itsystems@tdnet Jabber linux4miche...@jabber.ccc.de Owner Michelle Konzack Gewerbe Strasse 3 Tel office: +49-176-86004575 77694 Kehl Tel mobil: +49-177-9351947 Germany Tel mobil: +33-6-61925193 (France) USt-ID: DE 278 049 239 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
