[asterisk-users] BRI Installation
Hi Guys, All my installs are based on PRI ISDN. I now have a site that I need to install BRI. As I have not done a BRI install before I'm wanting to get some information from the people in the know if I need to do anything special. Typically I install libpri, dahdi Linux and tools, asterisk ...and then configure dahdi as one does for the required hardware. Is the same true for BRI with the exception of the libpri? I have this feeling that I need to install some other Linux drivers or something for BRI. I've purchased a Digium HB8 card and I don't see any mention of this in /etc/dahdi/modules. I've looked over the documentation at https://www.digium.com/en/supportcenter/documentation/viewdocs/H8 but there doesn't seem to be anything there that tells me how to configure dahdi or asterisk. If someone could give me some direction that would be greatly appreciated. Regards David-- _ -- 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] IMAP integration with MS Exchange 2010
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ric Marques
> Sent: Monday, June 04, 2012 3:51 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] IMAP integration with MS Exchange 2010
>
> Greetings,
>
> Has anybody here successfully integrated IMAP voicemail with Exchange 2010?
> If so - could you point me in the right direction to configure this
> properly?
>
> I'm running Asterisk 1.8.11-cert2.
>
> My voicemail.conf:
>
> [general]
> format=wav49
> imapserver=mail.domain.com
> authuser=asterisk_master
> authpassword=asterisk_master_password
> expungeonhangup=no
> pollmailboxes=yes
> pollfreq=30
> imapgreetings=no
> userscontext=customercontext
> searchcontexts=yes
>
> (*I have tried adding imapport=993 and imapflags=ssl, but just adds a
> 'security problem' error message to the logfiles).
>
> my voicemail users are in Realtime, and I have a column populated and set
> for imapuser (you can see in the log errors below that we are successfully
> getting imapuser from the database and inserting it in the connect string).
>
> Here's what happens when I try to leave a voicemail message:
>
> [2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849
> mm_log: IMAP Warning: Can't use Kerberos: invalid /authuser
> [2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849
> mm_log: IMAP Warning: SECURITY PROBLEM: insecure server advertised
> AUTH=PLAIN
> [2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849
> mm_log: IMAP Warning: Retrying PLAIN authentication after AUTHENTICATE
> failed.
> [2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849
> mm_log: IMAP Warning: SECURITY PROBLEM: insecure server advertised
> AUTH=PLAIN
> [2012-06-04 15:34:49] WARNING[32327]: app_voicemail_imapstorage.c:2849
> mm_log: IMAP Warning: Retrying PLAIN authentication after AUTHENTICATE
> failed.
> [2012-06-04 15:34:49] WARNING[32327]: app_voicemail_imapstorage.c:2849
> mm_log: IMAP Warning: SECURITY PROBLEM: insecure server advertised
> AUTH=PLAIN
> [2012-06-04 15:34:49] ERROR[32327]: app_voicemail_imapstorage.c:2852
> mm_log: IMAP Error: Can not authenticate to IMAP server: AUTHENTICATE
> failed.
> [2012-06-04 15:34:49] WARNING[32327]: app_voicemail_imapstorage.c:2849
> mm_log: IMAP Warning: [CLOSED] IMAP connection broken (server response)
> [2012-06-04 15:34:49] ERROR[32327]: app_voicemail_imapstorage.c:2598
> init_mailstream: Can't connect to imap server
> {mail.domain.com:143/imap/authuser=asterisk_master/notls/user=username}INBO
> X
> [2012-06-04 15:34:49] ERROR[32327]: app_voicemail_imapstorage.c:2122
> __messagecount: Houston we have a problem - IMAP mailstream is NULL
> [2012-06-04 15:34:49] NOTICE[32327]: app_voicemail_imapstorage.c:6159
> leave_voicemail: Can not leave voicemail, unable to count messages
>
>
> Am I just headed down a dead end road?
>
> Regards,
>
> Ric Marques
>
Please disregard... the client didn't really configure the master user that he
provided to me as a master account...
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[asterisk-users] IMAP integration with MS Exchange 2010
Greetings,
Has anybody here successfully integrated IMAP voicemail with Exchange 2010? If
so - could you point me in the right direction to configure this properly?
I'm running Asterisk 1.8.11-cert2.
My voicemail.conf:
[general]
format=wav49
imapserver=mail.domain.com
authuser=asterisk_master
authpassword=asterisk_master_password
expungeonhangup=no
pollmailboxes=yes
pollfreq=30
imapgreetings=no
userscontext=customercontext
searchcontexts=yes
(*I have tried adding imapport=993 and imapflags=ssl, but just adds a 'security
problem' error message to the logfiles).
my voicemail users are in Realtime, and I have a column populated and set for
imapuser (you can see in the log errors below that we are successfully getting
imapuser from the database and inserting it in the connect string).
Here's what happens when I try to leave a voicemail message:
[2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849 mm_log:
IMAP Warning: Can't use Kerberos: invalid /authuser
[2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849 mm_log:
IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849 mm_log:
IMAP Warning: Retrying PLAIN authentication after AUTHENTICATE failed.
[2012-06-04 15:34:48] WARNING[32327]: app_voicemail_imapstorage.c:2849 mm_log:
IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[2012-06-04 15:34:49] WARNING[32327]: app_voicemail_imapstorage.c:2849 mm_log:
IMAP Warning: Retrying PLAIN authentication after AUTHENTICATE failed.
[2012-06-04 15:34:49] WARNING[32327]: app_voicemail_imapstorage.c:2849 mm_log:
IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[2012-06-04 15:34:49] ERROR[32327]: app_voicemail_imapstorage.c:2852 mm_log:
IMAP Error: Can not authenticate to IMAP server: AUTHENTICATE failed.
[2012-06-04 15:34:49] WARNING[32327]: app_voicemail_imapstorage.c:2849 mm_log:
IMAP Warning: [CLOSED] IMAP connection broken (server response)
[2012-06-04 15:34:49] ERROR[32327]: app_voicemail_imapstorage.c:2598
init_mailstream: Can't connect to imap server
{mail.domain.com:143/imap/authuser=asterisk_master/notls/user=username}INBOX
[2012-06-04 15:34:49] ERROR[32327]: app_voicemail_imapstorage.c:2122
__messagecount: Houston we have a problem - IMAP mailstream is NULL
[2012-06-04 15:34:49] NOTICE[32327]: app_voicemail_imapstorage.c:6159
leave_voicemail: Can not leave voicemail, unable to count messages
Am I just headed down a dead end road?
Regards,
Ric Marques
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[asterisk-users] Asterisk 10.5.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Turn off warning message when bind address is set to any. (Closes issue ASTERISK-19456. Reported by Michael L. Young) * --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines (Closes issue ASTERISK-19727. Reported by Ben Klang) * --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. (Closes issue ASTERISK-19708. Reported by mehdi Shirazi) * --- Fix recalled party B feature flags for a failed DTMF atxfer. (Closes issue ASTERISK-19383. Reported by lgfsantos) * --- Fix DTMF atxfer running h exten after the wrong bridge ends. (Closes issue ASTERISK-19717. Reported by Mario) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.5.0 Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 1.8.13.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.13.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.13.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Turn off warning message when bind address is set to any. (Closes issue ASTERISK-19456. Reported by Michael L. Young) * --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit machines (Closes issue ASTERISK-19727. Reported by Ben Klang) * --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply before disconnecting the call. (Closes issue ASTERISK-19708. Reported by mehdi Shirazi) * --- Fix recalled party B feature flags for a failed DTMF atxfer. (Closes issue ASTERISK-19383. Reported by lgfsantos) * --- Fix DTMF atxfer running h exten after the wrong bridge ends. (Closes issue ASTERISK-19717. Reported by Mario) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.13.0 Thank you for your continued support of Asterisk! -- _ -- 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] Sangoma D100 Transcoder Asterisk 1.6
- Original Message - > I have installed and configures this card in asterisk 1.6. When > trying to load the module codec_sangoma.so I see the following in > the asterisk log. > [2012-06-04 15:50:31] WARNING[18168] loader.c: Error loading module > 'codec_sangoma.so': /usr/lib/asterisk/modules/codec_sangoma.so: > undefined symbol: ast_config_load > [2012-06-04 15:50:31] WARNING[18168] loader.c: Module > 'codec_sangoma.so' could not be loaded. > Has anyone had a similar issue with this card or have any idea what > the undefined symbol: ast_config_load might mean to figure out what > direction to head for further debugging? It looks like maybe Wanpipe was not compiled against the same version of Asterisk/DAHDI you're running. That would be the first thing to check. Next stop, Sangoma support. They are fantastic, and support is free. --Tim -- _ -- 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] Sangoma D100 Transcoder Asterisk 1.6
I have installed and configures this card in asterisk 1.6. When trying to load the module codec_sangoma.so I see the following in the asterisk log. [2012-06-04 15:50:31] WARNING[18168] loader.c: Error loading module 'codec_sangoma.so': /usr/lib/asterisk/modules/codec_sangoma.so: undefined symbol: ast_config_load [2012-06-04 15:50:31] WARNING[18168] loader.c: Module 'codec_sangoma.so' could not be loaded. Has anyone had a similar issue with this card or have any idea what the undefined symbol: ast_config_load might mean to figure out what direction to head for further debugging? -- _ -- 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 over IP?
- Original Message - > Hi Tim, > ... > While the fax machine starts to send the fax after a while it gives > the message, 'Fax failed' with error code: '388'. Is it the end > point fax machine issue or else? Please assist me out to resolve > this issue at earliest. Please do not email me directly. I've already responded on list, despite wanting to let this sit for a few days in response to you asking for support 'at earliest'... The Asterisk support list has no SLA, only governed by the time and willingness of the members to participate. Thanks. --Tim -- _ -- 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] Automate SLA testing
Thanks Paul, we are looking into the testsuite. > Date: Sun, 3 Jun 2012 18:32:57 -0400 > From: paul.belan...@polybeacon.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Automate SLA testing > > On 12-06-03 03:56 PM, Matt Hamilton wrote: > > > > We would like to automate Shared Line Appearance testing (e.g. phoneA > > answers a call, puts in on hold, phoneB picks up the call on hold) in our > > lab. Are there any tools/SIP call generators/clients that may help us > > create such a scenario? > > > Check out the asterisk testsuite for some examples[1]. You could use a > combination of StarPy and pjsua (python bindings) to do this. > > [1] http://svnview.digium.com/svn/testsuite/asterisk/trunk/ > > -- > Paul Belanger | PolyBeacon, Inc. > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > Github: https://github.com/pabelanger | Twitter: > https://twitter.com/pabelanger > > -- > _ > -- 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 over IP?
- Original Message -
> Hi Tim,
> I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is
> T.38 and when I try to send the fax from a fax machine i.e. HP 3180,
> I'm getting some warnings as listed below;
> -- Executing [4112345678@default:1] Goto("SIP/192.168.1.69-0005",
> "fax-detect,fax,1") in new stack
> -- Goto (fax-detect,fax,1)
> -- Executing [fax@fax-detect:1] NoOp("SIP/192.168.1.69-0005",
> " FAX DETECTED ") in new stack
> -- Executing [fax@fax-detect:2] Goto("SIP/192.168.1.69-0005",
> "fax-receive,receive,1") in new stack
> -- Goto (fax-receive,receive,1)
> -- Executing [receive@fax-receive:1]
> NoOp("SIP/192.168.1.69-0005", " FAX RECEIVE ") in new
> stack
> -- Executing [receive@fax-receive:2] Set("SIP/192.168.1.69-0005",
> "GLOBAL(FAXCOUNT)=5") in new stack
> == Setting global variable 'FAXCOUNT' to '5'
> -- Executing [receive@fax-receive:3] Set("SIP/192.168.1.69-0005",
> "FAXCOUNT=5") in new stack
> -- Executing [receive@fax-receive:4] Set("SIP/192.168.1.69-0005",
> "FAXFILE=fax-5-rx.tif") in new stack
> -- Executing [receive@fax-receive:5] Set("SIP/192.168.1.69-0005",
> "GLOBAL(LASTFAXCALLERNUM)=6461234567") in new stack
> == Setting global variable 'LASTFAXCALLERNUM' to '6461234567'
> -- Executing [receive@fax-receive:6] Set("SIP/192.168.1.69-0005",
> "GLOBAL(LASTFAXCALLERNAME)=") in new stack
> == Setting global variable 'LASTFAXCALLERNAME' to ''
> -- Executing [receive@fax-receive:7]
> NoOp("SIP/192.168.1.69-0005", " SETTING FAXOPT ") in new
> stack
> -- Executing [receive@fax-receive:8] Set("SIP/192.168.1.69-0005",
> "FAXOPT(ecm)=yes") in new stack
> -- Executing [receive@fax-receive:9] Set("SIP/192.168.1.69-0005",
> "FAXOPT(headerinfo)=MY FAXBACK RX") in new stack
> -- Executing [receive@fax-receive:10]
> Set("SIP/192.168.1.69-0005",
> "FAXOPT(localstationid)=1234567890") in new stack
> -- Executing [receive@fax-receive:11]
> Set("SIP/192.168.1.69-0005", "FAXOPT(maxrate)=14400") in new
> stack
> -- Executing [receive@fax-receive:12]
> Set("SIP/192.168.1.69-0005", "FAXOPT(minrate)=2400") in new
> stack
> -- Executing [receive@fax-receive:13]
> NoOp("SIP/192.168.1.69-0005", "FAXOPT(ecm) : yes") in new stack
> -- Executing [receive@fax-receive:14]
> NoOp("SIP/192.168.1.69-0005", "FAXOPT(headerinfo) : MY FAXBACK
> RX") in new stack
> -- Executing [receive@fax-receive:15]
> NoOp("SIP/192.168.1.69-0005", "FAXOPT(localstationid) :
> 1234567890") in new stack
> -- Executing [receive@fax-receive:16]
> NoOp("SIP/192.168.1.69-0005", "FAXOPT(maxrate) : 14400") in new
> stack
> -- Executing [receive@fax-receive:17]
> NoOp("SIP/192.168.1.69-0005", "FAXOPT(minrate) : 2400") in new
> stack
> -- Executing [receive@fax-receive:18]
> NoOp("SIP/192.168.1.69-0005", " RECEIVING FAX : fax-5-rx.tif
> ") in new stack
> -- Executing [receive@fax-receive:19]
> ReceiveFAX("SIP/192.168.1.69-0005",
> "/var/spool/asterisk/fax/fax-5-rx.tif") in new stack
> -- Channel 'SIP/192.168.1.69-0005' receiving FAX
> '/var/spool/asterisk/fax/fax-5-rx.tif'
> == Using UDPTL CoS mark 5
> [Jun 4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG
> detected but no fax extension
> [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init:
> channel 'SIP/192.168.1.69-0005' refused to negotiate T.38
> [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init:
> Audio FAX not allowed on channel 'SIP/192.168.1.69-0005' and
> T.38 negotiation failed; aborting.
> [Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error
> initializing channel 'SIP/192.168.1.69-0005' in T.38 mode
> == Spawn extension (fax-receive, receive, 19) exited non-zero on
> 'SIP/192.168.1.69-0005'
> In my sip.conf global configuration I enabled 'fax detect' and
> 't38pt_udptl' and added Cisco VGW peer;
> [CiscoVGW-10.70.X.X]
> host=10.70.X.X
> type=friend
> disallow=all
> allow=ulaw
> allow=alaw
> nat=yes
> insecure=port,invite
> context=fax-call
> canreinvite=no
> qualify=yes
> dtmfmode=inband
T.38 failed to negotiate. That means either your Asterisk side, or your Cisco
side are not playing nicely together. A packet capture of the call setup would
be helpful to determine which side is having the issues.
--Tim
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Re: [asterisk-users] Fax over IP?
Hi Tim,
I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is T.38
and when I try to send the fax from a fax machine i.e. HP 3180, I'm getting
some warnings as listed below;
-- Executing [4112345678@default:1] Goto("SIP/192.168.1.69-0005",
"fax-detect,fax,1") in new stack
-- Goto (fax-detect,fax,1)
-- Executing [fax@fax-detect:1] NoOp("SIP/192.168.1.69-0005", "
FAX DETECTED ") in new stack
-- Executing [fax@fax-detect:2] Goto("SIP/192.168.1.69-0005",
"fax-receive,receive,1") in new stack
-- Goto (fax-receive,receive,1)
-- Executing [receive@fax-receive:1] NoOp("SIP/192.168.1.69-0005",
" FAX RECEIVE ") in new stack
-- Executing [receive@fax-receive:2] Set("SIP/192.168.1.69-0005",
"GLOBAL(FAXCOUNT)=5") in new stack
== Setting global variable 'FAXCOUNT' to '5'
-- Executing [receive@fax-receive:3] Set("SIP/192.168.1.69-0005",
"FAXCOUNT=5") in new stack
-- Executing [receive@fax-receive:4] Set("SIP/192.168.1.69-0005",
"FAXFILE=fax-5-rx.tif") in new stack
-- Executing [receive@fax-receive:5] Set("SIP/192.168.1.69-0005",
"GLOBAL(LASTFAXCALLERNUM)=6461234567") in new stack
== Setting global variable 'LASTFAXCALLERNUM' to '6461234567'
-- Executing [receive@fax-receive:6] Set("SIP/192.168.1.69-0005",
"GLOBAL(LASTFAXCALLERNAME)=") in new stack
== Setting global variable 'LASTFAXCALLERNAME' to ''
-- Executing [receive@fax-receive:7] NoOp("SIP/192.168.1.69-0005",
" SETTING FAXOPT ") in new stack
-- Executing [receive@fax-receive:8] Set("SIP/192.168.1.69-0005",
"FAXOPT(ecm)=yes") in new stack
-- Executing [receive@fax-receive:9] Set("SIP/192.168.1.69-0005",
"FAXOPT(headerinfo)=MY FAXBACK RX") in new stack
-- Executing [receive@fax-receive:10] Set("SIP/192.168.1.69-0005",
"FAXOPT(localstationid)=1234567890") in new stack
-- Executing [receive@fax-receive:11] Set("SIP/192.168.1.69-0005",
"FAXOPT(maxrate)=14400") in new stack
-- Executing [receive@fax-receive:12] Set("SIP/192.168.1.69-0005",
"FAXOPT(minrate)=2400") in new stack
-- Executing [receive@fax-receive:13] NoOp("SIP/192.168.1.69-0005",
"FAXOPT(ecm) : yes") in new stack
-- Executing [receive@fax-receive:14] NoOp("SIP/192.168.1.69-0005",
"FAXOPT(headerinfo) : MY FAXBACK RX") in new stack
-- Executing [receive@fax-receive:15] NoOp("SIP/192.168.1.69-0005",
"FAXOPT(localstationid) : 1234567890") in new stack
-- Executing [receive@fax-receive:16] NoOp("SIP/192.168.1.69-0005",
"FAXOPT(maxrate) : 14400") in new stack
-- Executing [receive@fax-receive:17] NoOp("SIP/192.168.1.69-0005",
"FAXOPT(minrate) : 2400") in new stack
-- Executing [receive@fax-receive:18] NoOp("SIP/192.168.1.69-0005",
" RECEIVING FAX : fax-5-rx.tif ") in new stack
-- Executing [receive@fax-receive:19]
ReceiveFAX("SIP/192.168.1.69-0005",
"/var/spool/asterisk/fax/fax-5-rx.tif") in new stack
-- Channel 'SIP/192.168.1.69-0005' receiving FAX
'/var/spool/asterisk/fax/fax-5-rx.tif'
== Using UDPTL CoS mark 5
[Jun 4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG detected
but no fax extension
[Jun 4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init:
channel 'SIP/192.168.1.69-0005' refused to negotiate T.38
[Jun 4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init: Audio
FAX not allowed on channel 'SIP/192.168.1.69-0005' and T.38 negotiation
failed; aborting.
[Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error
initializing channel 'SIP/192.168.1.69-0005' in T.38 mode
== Spawn extension (fax-receive, receive, 19) exited non-zero on
'SIP/192.168.1.69-0005'
In my sip.conf global configuration I enabled 'fax detect' and
't38pt_udptl' and added Cisco VGW peer;
[CiscoVGW-10.70.X.X]
host=10.70.X.X
type=friend
disallow=all
allow=ulaw
allow=alaw
nat=yes
insecure=port,invite
context=fax-call
canreinvite=no
qualify=yes
dtmfmode=inband
While the fax machine starts to send the fax after a while it gives the
message, 'Fax failed' with error code: '388'. Is it the end point fax
machine issue or else? Please assist me out to resolve this issue at
earliest.
>
>
> > Thanks for your response. Here is my topology as listing down below;
>
> > PSTN Line --> Cisco Voice GW --> IP Cloud --> Asterisk
>
> > Will Asterisk able to receive the fax (as in topology above) using
> > its' fax module? In sip.conf I enabled fax detection and T.38.
> > Actually I don't want
> > to use Hylafax + iaxmodem as per requirement.
>
> If your Cisco voice gateway can deliver the calls using T.38, that should
> give you decent reliability. You'll want to us Asterisk 10 which has the
> best T.38 support at this point (compared to older releases). The receiving
> side of the equation then becomes whether to use Fax for Asterisk
> (commercial, 1 free channel, 2+ paid), or the included SpanDSP based fax
> module.
>
> --Tim
>
>
--
Regards,
[asterisk-users] Messaging capable app_sms
Hi, has someone done a patch for app_sms so it is capable of sending the SMS down the messaging-infrastructure in asterisk-10? Regards, Jay -- _ -- 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 OpenSMS API?
What do you want to do? Sending and receiving SMS? Am 03.06.2012 11:20, schrieb Michelle Konzack: Hello Experts, since connecting of 4 Huawei K3765-HV Sticks to my Server does not work, I now use the Vodafone EasyBox 803A (cost less then 30 Euro on eBay) and connect them to my ISDN cards. It has the advantage, that I can use in the same time the UMTS Internet connectivity and local analog telephones. However, if I use Windows, the program shiped with the EasyBox use the OpenSMS API to get the SMS from the USB-Stick trough the EasyBox. So, my questions are: 1) Does Asterisk (or an AddOn/PlugIn) support the OpenSMS API? 2) Does someone know, where I can get infos about the OpenSMS API? I have found nothing on Google. Thanks, Greetings and nice Day/Evening Michelle Konzack -- _ -- 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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
