Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
They used to have solaris on the Digium FTP site but they seem to be gone now :( On the free codec site they have some complied with icc and others with gcc4 so I don't see why you can't get this working with gcc on solaris. On Jan 15, 2008 4:01 AM, Bruce McAlister [EMAIL PROTECTED] wrote: Steve Totaro wrote: I would suggest building it yourself (http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt). It is not that difficult and ensures that it should be compatible with your machine. Just a little work. Has anyone tried building this on Solaris, I just had a look at the link and it looks like the Intel IPP stuff is only released for Windows, Linux and MAC. And the v32 G729 codec from Digium does not load within asterisk on Solaris, sooo, the Solaris users out there dont have much support when it comes to G729 codecs, a real pity really, this stops some large scale roll-outs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Andrew Joakimsen wrote: On Jan 14, 2008 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: Anyways, buying the license is the right thing to do unless you live where software patent laws are not applicable. Totally agree. I have bought many more licenses from asterisk than I've ever used, and mostly use the asterisk.hosting.lv codecs. Twice now while using the digium codec, upon upgrading asterisk, it stopped working. The Beta codec (based on IPP5), is much much faster than either the digium or the older codec, and at home (only place I run beta software), there hasn't been a problem. Mind you, according to show translation, the older codec (based on IPP4), is faster than the digium codec too. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Steve Totaro wrote: I would suggest building it yourself (http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt). It is not that difficult and ensures that it should be compatible with your machine. Just a little work. Has anyone tried building this on Solaris, I just had a look at the link and it looks like the Intel IPP stuff is only released for Windows, Linux and MAC. And the v32 G729 codec from Digium does not load within asterisk on Solaris, sooo, the Solaris users out there dont have much support when it comes to G729 codecs, a real pity really, this stops some large scale roll-outs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Andrew Joakimsen wrote: They used to have solaris on the Digium FTP site but they seem to be gone now :( On the free codec site they have some complied with icc and others with gcc4 so I don't see why you can't get this working with gcc on solaris. If you can, be sure to submit it to [EMAIL PROTECTED] , I'm sure he'll be happy to receive it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme recording
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
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Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Andrew Joakimsen wrote: They used to have solaris on the Digium FTP site but they seem to be gone now :( On the free codec site they have some complied with icc and others with gcc4 so I don't see why you can't get this working with gcc on solaris. Digium do still have the Solaris version of their codec on their download site and the following url: http://downloads.digium.com/pub/telephony/codec_g729/unsupported/ This codec is at version 32, whereas the latest is at 33. We tried this codec with valid licenses too, but the codec just fails to load in Asterisk. I was under the impression that the free codec required the Intel IPP libraries to be available on the system, or, statically linked into the codec. How would one build the codec if you could not link in a Solaris version of the IPP libraries, or am I missing something fundamental here? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.17 crashing more
Where we can see the log when this crashed coming. after that we can investigate for that particularly error. Before 1.4.17 we was using 1.2.X but we faced problem call hanged on console for one day and two day without any media and RTP. Once the call removed it comes in our billing with high duration which damage whole balance of customers for this reason we came to use 1.4.17 and till now we did not get such issue. Thank You Hi All, We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it stop to response to the SIP Clinets so they cannot make call or register. But safe_asterisk not restarting it back because asterisk running without any response to the sip clients. When we try to do 'core show channels' using Manager it returns only Action: Command Command: show channels That time asterisk not responding anything for clients for registration either for invitation. Please advice us how we can fix this issue. Upgrade to Asterisk 1.2.X unless you need the features in 1.4. Thanks, Steve Totaro - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
On Tue, Jan 15, 2008 at 09:05:35AM +, Thomas Kenyon wrote: Andrew Joakimsen wrote: On Jan 14, 2008 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: Anyways, buying the license is the right thing to do unless you live where software patent laws are not applicable. Totally agree. I have bought many more licenses from asterisk than I've ever used, and mostly use the asterisk.hosting.lv codecs. Just pointing out the obvious: the license you bought is not for using a general g729 codec. It is for one specific g729 codec, as distributed by Digium. Twice now while using the digium codec, upon upgrading asterisk, it stopped working. Yup. The extra costs of using non-free software. Don't use g729 if there's any other way. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme recording
On Tue, 15 Jan 2008, Lees, James (UK) wrote:
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Set the channel-variable: ${MEETME_RECORDINGFORMAT}
Described in doc/README.variables.
Use the command:
CLI show file formats
To list the formats avalable. Eg:
Format Name Extensions
g723 g723sf g723|g723sf
ulaw au au
slin slnsln|raw
ilbc iLBC ilbc
g726 g726-16g726-16
g726 g726-24g726-24
g726 g726-32g726-32
g726 g726-40g726-40
h263 h263 h263
alaw alaw alaw|al
g729 g729 g729
ulaw pcmpcm|ulaw|ul|mu
adpcm voxvox
gsmwav49 WAV|wav49
slin wavwav
gsmgsmgsm
Gordon
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Re: [asterisk-users] Meetme recording
In article [EMAIL PROTECTED],
Lees, James (UK) [EMAIL PROTECTED] wrote:
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
There's a clue in the help text you quoted. Set the channel variable
MEETME_RECORDINGFORMAT to one of the extensions shown in the output
of show file formats, before calling MeetMe. e.g.
exten = _X.,1,Set(MEETME_RECORDINGFORMAT=gsm)
exten = _X.,n,MeetMe(${EXTEN},dr)
Hope this helps!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Console app
Hi all I build an Asterisk, with asterisk 1.4.16.1 source. I have notice, that the console app don't appear on CLI... Is theres some options to turn on, when I compile asterisk? Thanks... -- Gilberto Nunes Itajaí - SC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
On Tue, Jan 15, 2008 at 11:08:33AM +, Thomas Kenyon wrote: If there was an equivalent free codec that provided good quality audio with such high compression and was widely supported, then I'd use it. Help make speex widely supported. Or continue to suffer with g729 and g723. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Tzafrir Cohen wrote: On Tue, Jan 15, 2008 at 09:05:35AM +, Thomas Kenyon wrote: Andrew Joakimsen wrote: On Jan 14, 2008 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: Anyways, buying the license is the right thing to do unless you live where software patent laws are not applicable. Totally agree. I have bought many more licenses from asterisk than I've ever used, and mostly use the asterisk.hosting.lv codecs. Just pointing out the obvious: the license you bought is not for using a general g729 codec. It is for one specific g729 codec, as distributed by Digium. I know, I just got tired of the trouble I was having with the digium codec. Twice now while using the digium codec, upon upgrading asterisk, it stopped working. Yup. The extra costs of using non-free software. Don't use g729 if there's any other way. Problem is, it seems to be good at what it does. If there was an equivalent free codec that provided good quality audio with such high compression and was widely supported, then I'd use it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme recording
Just set the variable ${MEETME_RECORDINGFORMAT} to the desired format
and voila its done.
Have fun.
on Tuesday 01/15/2008 Lees, James (UK)([EMAIL PROTECTED]) wrote
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
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How do
you spend it?
John Covici
[EMAIL PROTECTED]
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[asterisk-users] Fax machine detect
Hi, I was recently trying out with AMD (Answering Machine Detect) to detect the status of my call if it being picked up by HUMAN or MACHINE. Just want to know if any supporting features in asterisk 1.4.11 to detect if the call enters the Fax machine. Please provide the documentation link if any one has ideas on the same. Appreciate your response. Regards, Naveen.Palani Quinnox Consultancy Services Ltd | Pune | INDIA | Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 | Fax : +91 20 4015 2305 | Email : [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] “Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.” This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus. For more details about our company, visit http://www.Quinnox.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID blocking ...
J. Oquendo wrote: Hey all, when you guys have requests from clients to block their CID from showing through, what are others doing? I had a coworker throw in some Name Here0 garbage which none my carriers like. I don't want to do Private12345678910 so any suggestions. For SIP it should be Set(CALLERID(all)=Anonymous anonymous); per RFC 2543. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
On Jan 15, 2008 12:57 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Jan 14, 2008 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: I would argue that it is illegal. The main definition of illegal is 1. against law: contravening a specific law, especially a criminal law. http://encarta.msn.com/dictionary_/illegal.html Illegal means that something violates a criminal law. You linked to a page that describe the law in the US regarding patentholders registration of said patents. I'm not saying we should infringe on the patentholder's right I am simply saying it is not a criminal act, at least in the US. While it may not be against criminal law in the US it can be in France and Austria, in the US it is certainly against a specific law. http://en.wikipedia.org/wiki/Patent_law#Law Software is generally not patentable in the European Union (and probably in the countries that are pseudo-EU members) Anyways, buying the license is the right thing to do unless you live where software patent laws are not applicable. Totally agree. http://lists.digium.com/mailman/listinfo/asterisk-users Did you even bother to read the definition of against the law that I posted? In that definition, against law: contravening a specific law,, that being violating patent law. Then it goes on to say especially a criminal law Sorry Andrew, but I take Encarta's definition over yours. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Tzafrir Cohen wrote: On Tue, Jan 15, 2008 at 11:08:33AM +, Thomas Kenyon wrote: If there was an equivalent free codec that provided good quality audio with such high compression and was widely supported, then I'd use it. Help make speex widely supported. Or continue to suffer with g729 and g723. The problem with speex though is that for the same bit rate, the quality isn't as good as G.729, The transcode takes twice as much runtime (according to show translation recalc 10). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cisco ip phne 7911G with asterisk
hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I thought that the phone should ask the SEPmac.xml.cnf file but it asks CTLSEPmac.tlv all the time. I don't have this file in the server and it tries to download every few seconds whitout asking another file. According to what I have read this file shouldn't be neccesary and, when the phone cann't obtain it, the phone should ask SEPmac.xml.cnf. I don't know if I'm doing something bad or if it could be a issue of the firmware version. I would thank some clue. Thanks, Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 __ Web Revelación Yahoo! 2007: Premio Favorita del Público. http://es.promotions.yahoo.com/revelacion2007/favoritos/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Reason
Hello, I'm sniffing traffic between Asterisk and a Softswitch. I see that, in Decline SIP packages, there is a header called Reason and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1.4 and 1.2, but if it's only with Asterisk 1.4 will not be a big problem. Thank you, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playing DTMF tones down a channel
Hello, I am trying to play DTMF tones across a phone line to control the voicemail application. The voicemail app is not however detecting the tones. Does anyone have any suggestion of what I can change to help things along? I have tested playing the tones between two standard clients and the tones can be heard on the receiving client so the issue must be regarding the detection of the tone. Many thanks J This email and any attachments are confidential to the intended recipient and may also be privileged. If you are not the intended recipient please delete it from your system and notify the sender. You should not copy it or use it for any purpose nor disclose or distribute its contents to any other person. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] busy/congestion random
Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn pmp_l1_check=no nodialtone=no usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 context=from-pstn channel=1-2 channel=4-5 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming #include zapata-auto.conf group=1 context=from-pstn channel=1-2 channel=4-5 #include zapata_additional.conf #include zapata-BRI-HFC.conf ..the log is: Executing Macro(SIP/206-090a7dd8, dialout-trunk|1|348241||) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_TRUNK=1) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_NUMBER=348241) in new stack -- Executing Set(SIP/206-090a7dd8, ROUTE_PASSWD=) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?noauth) in new stack -- Goto (macro-dialout-trunk,s,6) -- Executing GotoIf(SIP/206-090a7dd8, 0?disabletrunk|1) in new stack -- Executing Set(SIP/206-090a7dd8, _NODEST=) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_TRUNK_OPTIONS=tT) in new stack -- Executing Set(SIP/206-090a7dd8, GROUP()=OUT_1) in new stack -- Executing Macro(SIP/206-090a7dd8, user-callerid|SKIPTTL) in new stack -- Executing NoOp(SIP/206-090a7dd8, user-callerid: device 206) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSER=206) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?report) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?start) in new stack -- Executing Set(SIP/206-090a7dd8, REALCALLERIDNUM=206) in new stack -- Executing NoOp(SIP/206-090a7dd8, REALCALLERIDNUM is 206) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSER=206) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSERCIDNAME=Centralino) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?report) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSERCID=206) in new stack -- Executing Set(SIP/206-090a7dd8, CALLERID(all)=Centralino 206) in new stack -- Executing Set(SIP/206-090a7dd8, REALCALLERIDNUM=206) in new stack -- Executing NoOp(SIP/206-090a7dd8, TTL: ARG1: SKIPTTL) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing NoOp(SIP/206-090a7dd8, Using CallerID Centralino 206) in new stack -- Executing Macro(SIP/206-090a7dd8, record-enable|206|OUT) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/206-090a7dd8, recordingcheck|20080115-131850|asterisk-12308-1200399530.1395) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20080115-131850|asterisk-12308-1200399530.1395: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/206-090a7dd8, No recording needed) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?skipoutcid) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_TRUNK_OPTIONS=tT) in new stack -- Executing Macro(SIP/206-090a7dd8, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/206-090a7dd8, REALCALLERIDNUM is 206) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set(SIP/206-090a7dd8, USEROUTCID=) in new stack -- Executing Set(SIP/206-090a7dd8, EMERGENCYCID=) in new stack -- Executing Set(SIP/206-090a7dd8, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf(SIP/206-090a7dd8, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,18) -- Executing GotoIf(SIP/206-090a7dd8, 1?report) in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp(SIP/206-090a7dd8, CallerID set to Centralino 206) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?nomax) in new stack -- Goto (macro-dialout-trunk,s,17) -- Executing AGI(SIP/206-090a7dd8, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/206-090a7dd8, OUTNUM=348241) in new stack -- Executing Set(SIP/206-090a7dd8, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?gocall) in new stack -- Goto
Re: [asterisk-users] SIP Reason
15 jan 2008 kl. 14.01 skrev Carles Pina i Estany: Hello, I'm sniffing traffic between Asterisk and a Softswitch. I see that, in Decline SIP packages, there is a header called Reason and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1.4 and 1.2, but if it's only with Asterisk 1.4 will not be a big problem. There's currently no way to access that header in Asterisk. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Reason
Hello, On Jan/15/2008, Johansson Olle E wrote: I'm sniffing traffic between Asterisk and a Softswitch. I see that, in Decline SIP packages, there is a header called Reason and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1.4 and 1.2, but if it's only with Asterisk 1.4 will not be a big problem. There's currently no way to access that header in Asterisk. Then... next days will be time to code :-D Thanks, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Reason
Won't SIP_HEADER(reason) do that for you?
e.g.
exten = 1996,1,Answer
exten = 1996,n,Set(sip_reason=${SIP_HEADER(reason)})
exten = 1996,n,NoOp(sip_reason)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Johansson Olle E
Sent: 15 January 2008 13:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Reason
15 jan 2008 kl. 14.01 skrev Carles Pina i Estany:
Hello,
I'm sniffing traffic between Asterisk and a Softswitch. I
see that, in
Decline SIP packages, there is a header called Reason
and I would
like to access to the content of this header from Asterisk.
How I can access to Reason header content?
I would like to access here using ASterisk 1.4 and 1.2, but if it's
only with Asterisk 1.4 will not be a big problem.
There's currently no way to access that header in Asterisk.
/O
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[asterisk-users] sip channel error - extension pattern matching problem
Hi,
When I have the following extension matching defined:
exten = _an_.,1,NoOp(-- Context routing-sip-announcement for ${EXTEN} --)
Asterisk doesn't find it when it receives such SIP request:
--- SIP read from 192.168.129.38:7160 ---
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:192.168.129.38:7160;lr=on
...
for instance when I use
such extension:
exten = _vm_.,1,NoOp(-- Context routing-sip-voicemail for ${EXTEN} --)
Asterisk finds extensions for RURI like:
--- SIP read from 192.168.129.38:7160 ---
INVITE sip:[EMAIL PROTECTED] SIP/2.0
...
Is this an error?
What did I miss off not?
Thanks in advance
Tomasz
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Re: [asterisk-users] SIP Reason
15 jan 2008 kl. 15.39 skrev Steve Langstaff: Won't SIP_HEADER(reason) do that for you? No, that's only works on the INVITE that opens the dialog. The reason header comes in a reply. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
A new article in my Asterisk 1.4 series cover blinking lamps on SIP business phones. Read it to learn all the new things! http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ Regards, /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
On Tue, 2008-01-15 at 16:41 +0100, Johansson Olle E wrote: A new article in my Asterisk 1.4 series cover blinking lamps on SIP business phones. Read it to learn all the new things! http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ Nice one Olle. Before I possibly waste my time trying this does this blinkety lights magic also work with SCCP phones? Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
On 1/15/08, Johansson Olle E [EMAIL PROTECTED] wrote: A new article in my Asterisk 1.4 series cover blinking lamps on SIP business phones. Read it to learn all the new things! http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ I wonder - when this will be available from Realtime.. Managing more than 50 users makes static config a nightmare, and AFAIK there is no ways how to create hints with variables/extension masks. So, it is logical to ask for hint support in Realtime. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing DTMF tones down a channel
In article [EMAIL PROTECTED], Lees, James (UK) [EMAIL PROTECTED] wrote: I am trying to play DTMF tones across a phone line to control the voicemail application. The voicemail app is not however detecting the tones. Does anyone have any suggestion of what I can change to help things along? I have tested playing the tones between two standard clients and the tones can be heard on the receiving client so the issue must be regarding the detection of the tone. It all depends on: 1) How your Asterisk box is interfacing to the PSTN, and 2) How you are attempting to send the tones. Please give more details, and hopefully we can help you. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Reason
Sent: 15 January 2008 15:23 by Johansson Olle E 15 jan 2008 kl. 15.39 skrev Steve Langstaff: Won't SIP_HEADER(reason) do that for you? No, that's only works on the INVITE that opens the dialog. The reason header comes in a reply. Thanks Olle. At least no one else saw my foolishness :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
Atis Lezdins wrote: On 1/15/08, Johansson Olle E [EMAIL PROTECTED] wrote: A new article in my Asterisk 1.4 series cover blinking lamps on SIP business phones. Read it to learn all the new things! http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/ I wonder - when this will be available from Realtime.. Managing more than 50 users makes static config a nightmare, and AFAIK there is no ways how to create hints with variables/extension masks. So, it is logical to ask for hint support in Realtime. I fail to see the problem. An #exec in the dialplan and a custom script which reads your database and generates the hints does the trick. Maybe a bit too complicated for the newbie but feasible for larger installations. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
I wonder - when this will be available from Realtime.. Managing more than 50 users makes static config a nightmare, and AFAIK there is no ways how to create hints with variables/extension masks. So, it is logical to ask for hint support in Realtime. AFAIK hints are supported in Realtime: Set the priority as -1. Set the app as the hint. Regards Cameron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Call Recording
Are there any tricks to getting combine_wave to make? [EMAIL PROTECTED] combine_wave-0.3]# ls -al total 84 drwxr-xr-x 2 root root 4096 Jan 15 10:54 . drwxr-x--- 6 root root 4096 Jan 15 10:54 .. -rw-r--r-- 1 root root 351 Oct 6 2005 CHANGES -rw-r--r-- 1 root root 1123 Oct 6 2005 combine_wave-0.3.lsm -rw-r--r-- 1 root root 23280 Oct 6 2005 combine_wave.c -rw-r--r-- 1 root root 449 Oct 6 2005 combine_wave.h -rw-r--r-- 1 root root 1048 Oct 6 2005 combine_wave.man -rw-r--r-- 1 root root 17976 Oct 6 2005 LICENSE -rw-r--r-- 1 root root 459 Oct 6 2005 Makefile -rw-r--r-- 1 root root 341 Oct 6 2005 README -rw-r--r-- 1 root root 762 Oct 6 2005 wave_header.h [EMAIL PROTECTED] combine_wave-0.3]# nano README [EMAIL PROTECTED] combine_wave-0.3]# make gcc -O2 -Wall -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -c combine_wave.c combine_wave.c: In function ârunning_infoâ: combine_wave.c:22: error: missing terminating character combine_wave.c:24: error: âbâ undeclared (first use in this function) combine_wave.c:24: error: (Each undeclared identifier is reported only once combine_wave.c:24: error: for each function it appears in.) combine_wave.c:24: error: expected â)â before âtogglesâ combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: stray â\â in program combine_wave.c:24: error: missing terminating character combine_wave.c:36: error: expected â;â before â}â token combine_wave.c: In function âusageâ: combine_wave.c:42: error: missing terminating character combine_wave.c:44: error: âcombine_waveâ undeclared (first use in this function) combine_wave.c:44: error: âaâ undeclared (first use in this function) combine_wave.c:44: error: âdâ undeclared (first use in this function) combine_wave.c:44: error: expected â]â before âmilliâ combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: expected â)â before ânâ combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: stray â\â in program combine_wave.c:44: error: missing terminating character combine_wave.c:62: error: expected â;â before â}â token combine_wave.c: In function âstrsaveâ: combine_wave.c:71: warning: implicit declaration of function âstrlenâ combine_wave.c:71: warning: incompatible implicit declaration of built-in function âstrlenâ combine_wave.c:73: warning: implicit declaration of function âstrcpyâ combine_wave.c:73: warning: incompatible implicit declaration of built-in function âstrcpyâ combine_wave.c: In function âmainâ: combine_wave.c:604: warning: incompatible implicit declaration of built-in function âstrcpyâ combine_wave.c:991: warning: implicit declaration of function âmemcpyâ combine_wave.c:991: warning: incompatible implicit declaration of built-in function âmemcpyâ make: *** [combine_wave.o] Error 1 - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Steve Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 14, 2008 10:51 AM Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording You might take a few ideas from this combine.sh script which works for me. It uses the combine_wave program from http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame program to convert to mp3. It converts the entire directory /var/spool/asterisk/monitor/*-in.wav files to mp3 where the mp3 file doesn't already exist. S. File: combine.sh --- #!/bin/sh cd /var/spool/asterisk/monitor for f in *-in.wav do in=$f out=`echo $f | sed -e 's/-in.wav/-out.wav/'` tmpwav=`echo $f | sed -e 's/-in.wav/-both.wav/'` mp3=`echo $f | sed -e 's/-in.wav/.mp3/'` if [ -e $mp3 ] then continue fi # combine the two tracks into one stereo file /usr/local/bin/combine_wave -l $in -r $out -o $tmpwav 2/dev/null /usr/bin/lame --silent -h -b 96 $tmpwav $mp3 #
Re: [asterisk-users] SIP Reason
Hello, On Jan/15/2008, Steve Langstaff wrote: Sent: 15 January 2008 15:23 by Johansson Olle E 15 jan 2008 kl. 15.39 skrev Steve Langstaff: Won't SIP_HEADER(reason) do that for you? No, that's only works on the INVITE that opens the dialog. The reason header comes in a reply. Thanks Olle. At least no one else saw my foolishness :) I saw :-D and I spent some minutes sniffing the SIP conversation. Yes, I cannot access to reason header but I can access to From, as Johansson said :-) Beside code, any other way how to do it? -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_voicemail for spanish
Will do AK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: lunes, 14 de enero de 2008 11:48 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_voicemail for spanish No features are being added for 1.2 so I'd check to see if 1.4 has the changes you need before filing a bugreport. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Park() help, extension not heard
On Mon, 2008-01-14 at 21:02 -0800, Rob wrote: I can place a call between two internal extensions, then on one extension transfer the call to extension 700, and the call gets parked on 701 but I don't hear the extension number when I do the transfer. I can hangup and call 701 and get the call back. This was an unfortunate bug in the 1.4.17 release. It's since been corrected, and will be fixed in the 1.4.18 release. (You could also grab the latest version of the 1.4 branch from the Subversion repository, except it's currently down for maintenance.) -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax machine detect
Hi, I was recently trying out with AMD (Answering Machine Detect) to detect the status of my call if it being picked up by HUMAN or MACHINE. Just want to know if any supporting features in asterisk 1.4.11 to detect if the call enters the Fax machine. Please provide the documentation link if any one has ideas on the same. Appreciate your response. Regards, Naveen.Palani Quinnox Consultancy Services Ltd | Pune | INDIA | Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 | Fax : +91 20 4015 2305 | Email : [EMAIL PROTECTED]mailto:[EMAIL PROTECTED] “Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.” This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus. For more details about our company, visit http://www.Quinnox.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Call Recording
Hi Mike,
On Tue, 2008-01-15 at 10:57 -0600, Mike Hammett wrote:
Are there any tricks to getting combine_wave to make?
Patch attached. Builds fine with patch on Fedora 8.
Regards,
Patrick
diff -Naur combine_wave-0.3.orig/combine_wave.c combine_wave-0.3/combine_wave.c
--- combine_wave-0.3.orig/combine_wave.c 2005-10-06 14:44:10.0 +0200
+++ combine_wave-0.3/combine_wave.c 2007-10-05 21:02:17.0 +0200
@@ -19,8 +19,8 @@
void running_info()
{
-fprintf(stderr,\
-RUNNNING COMMANDS
+fprintf(stderr,
+RUNNNING COMMANDS\n\
b toggles move both channels / move right channel delay mode.\n\
ESC exits.\n\
'z' 'x' 1 sample forward / backward.\n\
@@ -39,8 +39,8 @@
void usage()
{
-fprintf(stderr,\
-Usage:
+fprintf(stderr,
+Usage:\n\
combine_wave [-a] [-d milli seconds delay right channel relative to left]\n\
[-e samples delay right channel relative to left]\n\
[-k] -l filename_left [-m] -o output_filename -r filename_right [s start seek offset].\n\
diff -Naur combine_wave-0.3.orig/combine_wave.h combine_wave-0.3/combine_wave.h
--- combine_wave-0.3.orig/combine_wave.h 2005-10-06 14:44:10.0 +0200
+++ combine_wave-0.3/combine_wave.h 2007-10-05 21:02:52.0 +0200
@@ -5,6 +5,7 @@
#include unistd.h
#include stdio.h
#include stdlib.h
+#include string.h
#include signal.h
#include errno.h
diff -Naur combine_wave-0.3.orig/Makefile combine_wave-0.3/Makefile
--- combine_wave-0.3.orig/Makefile 2005-10-06 14:44:10.0 +0200
+++ combine_wave-0.3/Makefile 2007-10-05 21:00:43.0 +0200
@@ -6,13 +6,13 @@
CFLAGS = -O2 -Wall -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
.c.o:
- gcc $(CFLAGS) -c $
+ $(CC) $(CFLAGS) -c $
OBJECT =\
combine_wave.o
a.out : $(OBJECT)
- gcc -o combine_wave $(OBJECT)
+ $(CC) $(LDFLAGS) -o combine_wave $(OBJECT)
# DEPENDENCIES
combine_wave.o : combine_wave.c combine_wave.h wave_header.h
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Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
On Tuesday 15 January 2008 10:51:31 CSB wrote: I wonder - when this will be available from Realtime.. Managing more than 50 users makes static config a nightmare, and AFAIK there is no ways how to create hints with variables/extension masks. So, it is logical to ask for hint support in Realtime. AFAIK hints are supported in Realtime: Set the priority as -1. Set the app as the hint. That's all well and good, but as the entry is in the database, it is not able to keep a pointer to a routine in memory (or else the pointer may become invalid after restart, without clearing the database, or else you have multiple servers, or .) Having a hint entry is only half the battle. The other half is in keeping a registry of function pointers to call when the state of the device changes. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interrupt the swift text
Hi,
I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift
text to audio.
My functionality is as for example i make this text to audio deliver the person
called.
Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 Press
1 to confirm. Press 3 to cancel.
extension.conf dialplan:
[dialout]
exten =
outbound-handler,1,Dial(SIP/102,60,gM(outbound-connect^agi://10.1.1.68/ivr/speak^${CallInitiate_hashdatamailto:SIP/[EMAIL
PROTECTED],60,gM(outbound-connect^agi://10.1.1.68/ivr/speak^${CallInitiate_hashdata}^${MACHINE_STATUS_UNKNOWN}))
[macro-outbound-connect]
exten = s,1,Answer()
exten = s,2,System(swift -o /tmp/test.wav -p
audio/channels=1,audio/sampling-rate=8000 Press 1 to confirm. Press 3 to
cancel.)
exten = s,3,Background(/tmp/test)
exten = s,4,Hangup
exten = 1,1,Playback(thanks)
exten = 2,1,Playback(bye)
Here in this, the call is connected and answered the control transfer to macro
context. One way i can interrupt the text before it completes the text is by
using 'Background (/tmp/test)' to play the audio.
When iam in the middle of the audio if i press 1 before it completes the entire
text, the control should go to 'exten = 1,1,Playback(thanks)'. But in macro
the 'Background' doesnt seem to work. It works fine outside macro context.
When i use the Asterisk cmd GoTo(new_context,extn,priority) inside macro, I get
a message 'channel jumping out of macro outbound-connect' waits for a minute
and hungs up, the control doesnt go to new_context.
Does anyone have any ideas i can work it out. How can i have the Asterisk cmd
Background inside macro? or how to execute the GoTo command?
Thanks and appreciate your response.
Regards,
Naveen.Palani
“Quinnox, a global IT services company prides itself on its SEI-CMM Level 5,
ISO‑9001:2000 assessed delivery processes and provides solutions in areas of
E-Business, ERP, Application Management Services, and EAI to customers in BFSI,
Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global
Delivery Model.”
This e-mail and any attached files are confidential, proprietary, and may also
be legally privileged information, and are intended solely for the use of the
individual or entity to whom they are addressed. If you are not the intended
recipient of this e-mail, please send it back to the person who sent it to you
and delete the e-mail and any attached files and destroy any copies of it; you
may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED]
Quinnox Consultancy Services and/or any of its sister companies owns no
responsibility for the views presented in the e-mail and any attached files
unless the sender mentions so, with due authority of Quinnox Consultancy
Services.
Unauthorized reading, reproduction, publication, use, dissemination,
forwarding, printing or copying of this e-mail and its attachments is
prohibited.
We have checked this message for any known viruses; however we decline any
liability, in case of any damage caused by a non-detected virus.
For more details about our company, visit http://www.Quinnox.com
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Re: [asterisk-users] Asterisk 1.4 Call Recording
I'm a newb when it comes to patch. I have a combine_wave-0.3.orig and a
combine_wave-0.3 directory. This is what I get:
[EMAIL PROTECTED] ~]# patch combine_wave-0.3.patch
can't find file to patch at input line 4
Perhaps you should have used the -p or --strip option?
The text leading up to this was:
--
|diff -Naur combine_wave-0.3.orig/combine_wave.c
combine_wave-0.3/combine_wave.c
|--- combine_wave-0.3.orig/combine_wave.c 2005-10-06
14:44:10.0 +0200
|+++ combine_wave-0.3/combine_wave.c2007-10-05 21:02:17.0 +0200
--
File to patch:
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 15, 2008 11:19 AM
Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording
Hi Mike,
On Tue, 2008-01-15 at 10:57 -0600, Mike Hammett wrote:
Are there any tricks to getting combine_wave to make?
Patch attached. Builds fine with patch on Fedora 8.
Regards,
Patrick
diff -Naur combine_wave-0.3.orig/combine_wave.c
combine_wave-0.3/combine_wave.c
--- combine_wave-0.3.orig/combine_wave.c 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.c 2007-10-05 21:02:17.0 +0200
@@ -19,8 +19,8 @@
void running_info()
{
-fprintf(stderr,\
-RUNNNING COMMANDS
+fprintf(stderr,
+RUNNNING COMMANDS\n\
b toggles move both channels / move right channel delay mode.\n\
ESC exits.\n\
'z' 'x' 1 sample forward / backward.\n\
@@ -39,8 +39,8 @@
void usage()
{
-fprintf(stderr,\
-Usage:
+fprintf(stderr,
+Usage:\n\
combine_wave [-a] [-d milli seconds delay right channel relative to
left]\n\
[-e samples delay right channel relative to left]\n\
[-k] -l filename_left [-m] -o output_filename -r filename_right [s start
seek offset].\n\
diff -Naur combine_wave-0.3.orig/combine_wave.h
combine_wave-0.3/combine_wave.h
--- combine_wave-0.3.orig/combine_wave.h 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.h 2007-10-05 21:02:52.0 +0200
@@ -5,6 +5,7 @@
#include unistd.h
#include stdio.h
#include stdlib.h
+#include string.h
#include signal.h
#include errno.h
diff -Naur combine_wave-0.3.orig/Makefile combine_wave-0.3/Makefile
--- combine_wave-0.3.orig/Makefile 2005-10-06 14:44:10.0 +0200
+++ combine_wave-0.3/Makefile 2007-10-05 21:00:43.0 +0200
@@ -6,13 +6,13 @@
CFLAGS = -O2 -Wall -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
.c.o:
- gcc $(CFLAGS) -c $
+ $(CC) $(CFLAGS) -c $
OBJECT =\
combine_wave.o
a.out : $(OBJECT)
- gcc -o combine_wave $(OBJECT)
+ $(CC) $(LDFLAGS) -o combine_wave $(OBJECT)
# DEPENDENCIES
combine_wave.o : combine_wave.c combine_wave.h wave_header.h
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Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
On 1/15/08, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 15 January 2008 10:51:31 CSB wrote: I wonder - when this will be available from Realtime.. Managing more than 50 users makes static config a nightmare, and AFAIK there is no ways how to create hints with variables/extension masks. So, it is logical to ask for hint support in Realtime. AFAIK hints are supported in Realtime: Set the priority as -1. Set the app as the hint. Oh, haven't seen anything like this, and not even any queries in log that asks for -1 priority Any more docs on this? That's all well and good, but as the entry is in the database, it is not able to keep a pointer to a routine in memory (or else the pointer may become invalid after restart, without clearing the database, or else you have multiple servers, or .) Having a hint entry is only half the battle. The other half is in keeping a registry of function pointers to call when the state of the device changes. I'm not very familiar with internal structure of device states, however i think this is not so hard. Why would you need to keep a registry, i think you should just do SELECT whenever any device state changes, to find out what to update. While on topic - may i ask for help.. For queues with dynamic members (i.e. Local/[EMAIL PROTECTED]), if i would create something like this (in realtime of course) [from-queue] exten=200,hint,SIP/300 exten=200,1,Dial(SIP/300) Would this send update to queue of SIP/300 state? All the RINGING/INUSE? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Call Recording
Never mind, I got it. I needed a -p0
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 15, 2008 11:19 AM
Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording
Hi Mike,
On Tue, 2008-01-15 at 10:57 -0600, Mike Hammett wrote:
Are there any tricks to getting combine_wave to make?
Patch attached. Builds fine with patch on Fedora 8.
Regards,
Patrick
diff -Naur combine_wave-0.3.orig/combine_wave.c
combine_wave-0.3/combine_wave.c
--- combine_wave-0.3.orig/combine_wave.c 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.c 2007-10-05 21:02:17.0 +0200
@@ -19,8 +19,8 @@
void running_info()
{
-fprintf(stderr,\
-RUNNNING COMMANDS
+fprintf(stderr,
+RUNNNING COMMANDS\n\
b toggles move both channels / move right channel delay mode.\n\
ESC exits.\n\
'z' 'x' 1 sample forward / backward.\n\
@@ -39,8 +39,8 @@
void usage()
{
-fprintf(stderr,\
-Usage:
+fprintf(stderr,
+Usage:\n\
combine_wave [-a] [-d milli seconds delay right channel relative to
left]\n\
[-e samples delay right channel relative to left]\n\
[-k] -l filename_left [-m] -o output_filename -r filename_right [s start
seek offset].\n\
diff -Naur combine_wave-0.3.orig/combine_wave.h
combine_wave-0.3/combine_wave.h
--- combine_wave-0.3.orig/combine_wave.h 2005-10-06 14:44:10.0
+0200
+++ combine_wave-0.3/combine_wave.h 2007-10-05 21:02:52.0 +0200
@@ -5,6 +5,7 @@
#include unistd.h
#include stdio.h
#include stdlib.h
+#include string.h
#include signal.h
#include errno.h
diff -Naur combine_wave-0.3.orig/Makefile combine_wave-0.3/Makefile
--- combine_wave-0.3.orig/Makefile 2005-10-06 14:44:10.0 +0200
+++ combine_wave-0.3/Makefile 2007-10-05 21:00:43.0 +0200
@@ -6,13 +6,13 @@
CFLAGS = -O2 -Wall -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
.c.o:
- gcc $(CFLAGS) -c $
+ $(CC) $(CFLAGS) -c $
OBJECT =\
combine_wave.o
a.out : $(OBJECT)
- gcc -o combine_wave $(OBJECT)
+ $(CC) $(LDFLAGS) -o combine_wave $(OBJECT)
# DEPENDENCIES
combine_wave.o : combine_wave.c combine_wave.h wave_header.h
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Re: [asterisk-users] Interrupt the swift text
On Jan 15, 2008 12:32 PM, Naveen Palani [EMAIL PROTECTED] wrote:
Hi,
I am using Asterisk-1.4.11 version to make outbound calls and deliver the
swift text to audio.
My functionality is as for example i make this text to audio deliver the
person called.
Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000
Press 1 to confirm. Press 3 to cancel.
extension.conf dialplan:
[dialout]
exten = outbound-handler,1,Dial(
SIP/102,60,gM(outbound-connect^agi://10.1.1.68/ivr/speak^${CallInitiate_hashdataSIP/[EMAIL
PROTECTED],60,gM%28outbound-connect%5Eagi://10.1.1.68/ivr/speak%5E$%7BCallInitiate_hashdata%7D%5E$%7BMACHINE_STATUS_UNKNOWN
}))
[macro-outbound-connect]
exten = s,1,Answer()
exten = s,2,System(swift -o /tmp/test.wav -p
audio/channels=1,audio/sampling-rate=8000 Press 1 to confirm. Press 3 to
cancel.)
exten = s,3,Background(/tmp/test)
exten = s,4,Hangup
exten = 1,1,Playback(thanks)
exten = 2,1,Playback(bye)
Here in this, the call is connected and answered the control transfer to
macro context. One way i can interrupt the text before it completes the text
is by using 'Background (/tmp/test)' to play the audio.
When iam in the middle of the audio if i press 1 before it completes the
entire text, the control should go to 'exten = 1,1,Playback(thanks)'. But
in macro the 'Background' doesnt seem to work. It works fine outside macro
context.
When i use the Asterisk cmd GoTo(new_context,extn,priority) inside macro,
I get a message 'channel jumping out of macro outbound-connect' waits for
a minute and hungs up, the control doesnt go to new_context.
Does anyone have any ideas i can work it out. How can i have the Asterisk
cmd Background inside macro? or how to execute the GoTo command?
Thanks and appreciate your response.
Regards,
*Naveen.Palani*
This may or may not help. It was a solution to what seems a similar
problem in another recent post.
Use Read to get the extension at the end with GotoIf on the variable checks
and Background with the context parameter set to the macro context.
Thanks,
Steve Totaro
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Re: [asterisk-users] Interrupt the swift text
On Tuesday 15 January 2008 12:32, Naveen Palani wrote: Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 Press 1 to confirm. Press 3 to cancel. Naveen, How about generating the wav files and storing them, then playing the wav's from the call tree, rather then re-generating the TTS every time. These seem to be static in nature. Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Heartbeat
Has anyone ever written asterisk logic to Heartbeat remote phone lines? Something that would dial out and see if a busy tone is encountered and take some sort of action? If not, any good ideas on how to do it? Obviously this would involve .call files. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Record calls then send them to users voicemail
Just wondering if this is possible: Make a call from a registered sip extension (Doesn't matter if it's internal or external) during the call press a key sequence let say *90 to start recording call. When the call ends the recording automagically goes to their voicemail. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 context
I am running asterisk 1.4 with Cisco Call manager. I made a context for it of course in sip.conf when the call comes in it does not seem to be obeying the context though. Only way I could get the call to answer was to put the phone number (cut and paste the same lines here) into the default context. Do incoming sip calls not obey the context? Very strange. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interrupt the swift text
Steve/Ron, I also did that. I created a wave file and stored in /tmp directory and then use Background cmd inside macro. But it doesnt seem to work. I saw from the forum to use Background with the context parameter set to the macro context. Used it in this way, suggest me if it is wrong: exten = s,3,Background(/tmp/test|outbound-connect) But still doesnt work. Please suggest me. Regards, Naveen - Original Message - From: Ron Joffe [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 15, 2008 12:33 PM Subject: Re: [asterisk-users] Interrupt the swift text On Tuesday 15 January 2008 12:32, Naveen Palani wrote: Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 Press 1 to confirm. Press 3 to cancel. Naveen, How about generating the wav files and storing them, then playing the wav's from the call tree, rather then re-generating the TTS every time. These seem to be static in nature. Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users “Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.” This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus. For more details about our company, visit http://www.Quinnox.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
On Tuesday 15 January 2008 12:13:46 Atis Lezdins wrote: On 1/15/08, Tilghman Lesher [EMAIL PROTECTED] wrote: Having a hint entry is only half the battle. The other half is in keeping a registry of function pointers to call when the state of the device changes. I'm not very familiar with internal structure of device states, however i think this is not so hard. Why would you need to keep a registry, i think you should just do SELECT whenever any device state changes, to find out what to update. No, you need the REVERSE direction. You need to know what code to notify when the device changes (the device does not need to know that its state changed -- the device is what *initiated* the state change). And that code is not always specifically pointing to a device. In some cases, the code is notifying an application that a device changed, which is why it's a function pointer callback that is registered to the device hint. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interrupt the swift text
Naveen Palani wrote: Does anyone have any ideas i can work it out. How can i have the Asterisk cmd Background inside macro? or how to execute the GoTo command? I have really started to wish for 2 new standard commands - BackgroundApp and SpeechBackgroundApp to be added to Asterisk just for this sort of situation. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4 context
On 14:00, Tue 15 Jan 08, Jerry Geis wrote: I am running asterisk 1.4 with Cisco Call manager. I made a context for it of course in sip.conf when the call comes in it does not seem to be obeying the context though. Only way I could get the call to answer was to put the phone number (cut and paste the same lines here) into the default context. Do incoming sip calls not obey the context? Very strange. Looks like a configuration issue in sip.conf. We really need some more info about your configuration before we can help you. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and ISDN-BRI support -- Solution
Quoting Jaap Winius [EMAIL PROTECTED]:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware, distro,
kernel, modules, versions, config files).
Thanks to the support I received here I now have a working system, so
I thought I'd show my appreciation by posting my configuration here
for anyone who's interested.
Telco: KPN Telecom (Netherlands)
ISDN hardware: HFC-S PCI card (Cologne chip).
OS: Debian GNU/Linux stable (etch)
Kernel: 2.6.18-5-k7 (for an AMD Athlon CPU)
Relevant links in /etc/apt/sources.list:
deb http://updates.xorcom.com/rapid etch main
deb-src http://updates.xorcom.com/rapid etch main
Relevant installed debian packages:
asterisk 1.4.14~dfsg-0.4849
asterisk-config 1.4.14~dfsg-0.4849
asterisk-doc 1.4.14~dfsg-0.4849
asterisk-sounds-main 1.4.14~dfsg-0.4849
zaptel1.4.7.xpp.r5178-2
zaptel-firmware 1.4.7.xpp.r5178-2
zaptel-modules-2.6.18-5-k71.4.7.xpp.r5178-2+2.6.18.dfsg.1-17 *
zaptel-source 1.4.7.xpp.r5178-2
*) Compiled from zaptel-source using the command
m-a a-i zaptel.
Note: All of these packages are from xorcom.com. Debian etch
provides v1.2 of the Asterisk and Zaptel packages, which I
found to be too problematic.
Relevant loaded modules:
xpp89088 0
vzaphfc24984 3
zaptel185956 10 xpp,vzaphfc
firmware_class 10048 0
crc_ccitt 2560 1 zaptel
Note: The zaptel-modules package includes both the older zaphfc
and the newer vzaphfc modules. If genzaptelconf -d is run, both
get loaded, which is confusing at best. Therefore, I opted to
remove the older zaphfc module. I'm not sure the xpp and
firmware_class modules are necessary either: they also get loaded,
but don't seem to cause any trouble. Finally, I've found that the
modules I do need don't work properly unless they get loaded with
the genzaptelconf -d command. I guess that it loads them with
some parameters.
/etc/asterisk/zapata.conf:
[trunkgroups]
[channels]
context=isdn-in
language=en
overlapdial=yes
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callerid=asreceived
rxgain=4.5
txgain=-3
callgroup=1
pickupgroup=1
pridialplan=unknown
prilocaldialplan=unknown
nationalprefix=0
internationalprefix=00
echocancel=yes
echotraining=100
echocancelwhenbridged=yes
faxdetect=incoming
immediate=no
group=1
switchtype=euroisdn
signalling=bri_cpe
channel=1-2
Note: I doubt all of these settings are absolutely necessary,
but this works for me.
Relevant parts of /etc/asterisk/extensions.conf:
[globals]
[general]
[isdn-in]
exten = isdn-in,1,Goto(0715134449,1)
exten = 0031715134449,1,Goto(0715134449,1)
exten = 0715134449,1,Dial(SIP/1000,30)
exten = 0715134449,n,Hangup()
[outgoing]
exten = _003171.,1,Dial(Zap/g1/${EXTEN},,r)
[internal]
exten = 1000,1,Verbose(1|Extension 1000)
exten = 1000,n,Dial(SIP/1000,30)
exten = 1000,n,Hangup()
[phones]
include = internal
include = outgoing
Note: In the dial command, Dial(Zap/g1/${EXTEN},,r), g1
corresponds to group=1 in /etc/asterisk/zapata.conf.
/etc/asterisk/indications.conf:
[general]
country=nl
[nl]
description = Netherlands
ringcadence = 1000,4000
dial = 425
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/250,0/250
callwaiting = 425/500,0/9500
dialrecall = 425/500,0/50
record = 1400/500,0/15000
info = 950/330,1400/330,1800/330,0/1000
stutter = 425/500,0/50
Some diagnostic information:
# cat /proc/zaptel/*
Span 1: ZTHFC1 HFC-S PCI A Zaptel Driver card 0 [TE] (MASTER) AMI/CCS
1 ZTHFC1/0/1 Clear (In use)
2 ZTHFC1/0/2 Clear (In use)
3 ZTHFC1/0/3 HDLCFCS (In use)
Note: These channels are (In use) because Asterisk is using them.
# cat /proc/interrupts
CPU0
0: 218203798IO-APIC-edge timer
6: 3IO-APIC-edge floppy
8: 1IO-APIC-edge rtc
9: 0 IO-APIC-level acpi
15:129IO-APIC-edge ide1
169: 95059844 IO-APIC-level skge
177: 10547626 IO-APIC-level libata
185: 0 IO-APIC-level uhci_hcd:usb1, uhci_hcd:usb2, ...
193: 3923488639 IO-APIC-level vzaphfc
201: 0 IO-APIC-level via82cxxx
NMI: 0
LOC: 218195472
ERR: 0
MIS:
[asterisk-users] inbound Audio problems probably not NAT related?
Hello all, Was hoping to get a sanity check along with a question. Below is the output from top run with normal defaults, except to show both CPU's, on a SuSE 10.2 box with Asterisk v1.4.15. top - 10:00:58 up 3 days, 5:54, 4 users, load average: 0.15, 0.05, 0.01 Tasks: 110 total, 2 running, 108 sleeping, 0 stopped, 0 zombie Cpu0 : 0.2%us, 0.2%sy, 0.0%ni, 97.3%id, 2.2%wa, 0.1%hi, 0.0%si, 0.0%st Cpu1 : 0.3%us, 0.0%sy, 0.0%ni, 99.6%id, 0.1%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 4052276k total, 2586128k used, 1466148k free, 389208k buffers Swap: 4200956k total,0k used, 4200956k free, 1929952k cached from show channels:(was the same before and after top was run) 12 active channels 6 active calls Would any of the guru's here say that this was good, bad, middle of the road, not enough info to tell? At the time I copied this there were 5 active calls in show channels. This server is exhibiting some strange behavior and I was starting to think it may be system overload. I find this hard to accept given the specs but, hey I don't know everything! some info from /proc/cpuinfo: vendor_id : AuthenticAMD cpu family : 15 model : 35 model name : Dual Core AMD Opteron(tm) Processor 180 stepping: 2 cpu MHz : 2411.130 cache size : 1024 KB some info from /proc/meminfo: MemTotal: 4052276 kB MemFree: 1469356 kB Buffers:388196 kB Cached:1927548 kB SwapCached: 0 kB Active: 893644 kB Inactive: 1523168 kB HighTotal: 0 kB HighFree:0 kB LowTotal: 4052276 kB LowFree: 1469356 kB SwapTotal: 4200956 kB SwapFree: 4200956 kB Dirty: 228 kB Writeback: 0 kB Hardware RAID 5 on-motherboard gigE connected through Cisco switch On inbound calls I lose the incoming audio after a couple minutes, outbound audio is always good, then after a while inbound audio magically starts up again. this happens on maybe 10% of calls at its worst. I have looked at the possibility of NAT issues and do not believe that to be the case. I have noticed that the memory usage climbs steadily but I believe that is the kernel as top show no process with more than 0.4% memory usage. Although when I rebooted (yes, an act of desperation) over the weekend the amount of calls with this problem dropped dramatically along with total memory usage which is slowly climbing again. Started at about 1gig on Saturday morning and is now at the 2.6gig shown above in top. This box typically does around 35,000 minutes of calls each month with a couple busy periods each day during weekdays. Normally no more than 10 to 12 calls at one time. provider--T1 to Cisco router--Asterisk--phones The router is doing NAT and routing all traffic from a specific IP to the asterisk box and dropping everything from any other IP. canreinvite is set to no on the sip trunk and all the phones. One thing that may be related is that when I ssh into this box it takes a full minute respond after the pass phrase is typed in. Could this be related or am I just grasping at straws? Any Ideas? -- JohnM begin:vcard fn:John Millican n:;John Millican adr:;;PO Box 9;Wentworth;NH;03282;US email;internet:[EMAIL PROTECTED] title:Director of Technology tel;work:603-764-9163 x-mozilla-html:FALSE url:www.sentinelcommunications.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Record calls then send them to users voicemail
Lookup the automon feature on features.conf . Best regards, On Jan 15, 2008 4:55 PM, Anciso, Roy [EMAIL PROTECTED] wrote: Just wondering if this is possible: Make a call from a registered sip extension (Doesn't matter if it's internal or external) during the call press a key sequence let say *90 to start recording call. When the call ends the recording automagically goes to their voicemail. Thanks *Roy Anciso* Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.b ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Heartbeat
Have a look at the new Digium list: Asterisk-HA, I think this thread makes more sense there. Best Regards, On Jan 15, 2008 4:42 PM, Jeremy Mann [EMAIL PROTECTED] wrote: Has anyone ever written asterisk logic to Heartbeat remote phone lines? Something that would dial out and see if a busy tone is encountered and take some sort of action? If not, any good ideas on how to do it? Obviously this would involve .call files. -- This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.b ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attended transfers manager or phone
Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real simple answer to. I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Originate is part of the solution, but what really I want is the phone being taken off-hook and then being able to dial the number without having to answer the dial-back first. 1. One solution, though an ugly one, would be using Originate, but use a phone that has some sort tcp/ip interface that allows for taking the phone off-hook. 2. A Better solution would be using a phone that allows dialling and taking the phone off-hook on-hook etc. via some tcp/ip interface. 3. Yet another solution, though I do not favour this one since I really don't want to maintain the sip phone code, would be programming a soft sip phone with all the bells and whistles and adding the switchboard functionality to that (name searching, status email so on and so forth. In the end all I need is just a software or hardware phone, sip/iax, which can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps status requests. If such a phone exists that would do the trick, the rest is manageable via the Asterisk Manager console. I'm guessing some people have messed with this problem before so I hope that someone has some information about this kind of thing :) Thank you in advance Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Ok, let's just agree to disagree and say that using patented software without a patent license is wrong What I am saying is you can be sued to the poorhouse but you won't be arrested and put in jail. On Jan 15, 2008 7:55 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Jan 15, 2008 12:57 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Jan 14, 2008 7:50 PM, Steve Totaro [EMAIL PROTECTED] wrote: I would argue that it is illegal. The main definition of illegal is 1. against law: contravening a specific law, especially a criminal law. http://encarta.msn.com/dictionary_/illegal.html Illegal means that something violates a criminal law. You linked to a page that describe the law in the US regarding patentholders registration of said patents. I'm not saying we should infringe on the patentholder's right I am simply saying it is not a criminal act, at least in the US. While it may not be against criminal law in the US it can be in France and Austria, in the US it is certainly against a specific law. http://en.wikipedia.org/wiki/Patent_law#Law Software is generally not patentable in the European Union (and probably in the countries that are pseudo-EU members) Anyways, buying the license is the right thing to do unless you live where software patent laws are not applicable. Totally agree. Did you even bother to read the definition of against the law that I posted? In that definition, against law: contravening a specific law,, that being violating patent law. Then it goes on to say especially a criminal law Sorry Andrew, but I take Encarta's definition over yours. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
Hello, I wonder - when this will be available from Realtime.. Managing more than 50 users makes static config a nightmare, and AFAIK there is no ways how to create hints with variables/extension masks. So, it is logical to ask for hint support in Realtime. AFAIK hints are supported in Realtime: Set the priority as -1. Set the app as the hint. I have a small question: other than a phone (ie. SIP/something), what else can I use as app? Can I handle the change via some custom code? TIA, -- Dott. Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
Andrea Spadaccini wrote: I have a small question: other than a phone (ie. SIP/something), what else can I use as app? Can I handle the change via some custom code? There are a number of things that can provide device state in Asterisk. That includes real devices such as SIP endpoints, or any other channel driver. However, it also includes things like monitoring the state of a space in parking, or the usage of a MeetMe conference. I have also written a small dialplan function which lets you create custom device states. A lot of people use this for things like having a light on the phone that reflects whether the agent is logged in or not. More information: http://asterisk.org/node/48325 http://asterisk.org/node/48360 -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
Patrick wrote: Nice one Olle. Before I possibly waste my time trying this does this blinkety lights magic also work with SCCP phones? IIRC, this feature is currently only supported in Asterisk trunk (soon to become Asterisk 1.6). -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'
Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
Ciao Russell, I have a small question: other than a phone (ie. SIP/something), what else can I use as app? Can I handle the change via some custom code? There are a number of things that can provide device state in Asterisk. That includes real devices such as SIP endpoints, or any other channel driver. However, it also includes things like monitoring the state of a space in parking, or the usage of a MeetMe conference. I have also written a small dialplan function which lets you create custom device states. A lot of people use this for things like having a light on the phone that reflects whether the agent is logged in or not. More information: http://asterisk.org/node/48325 http://asterisk.org/node/48360 Thanks a lot for the info, I already read the first article, and it's great to know that DEVSTATE can be used in 1.4. But my question was different, my poor english doesn't help me. :( In your article I read For example, when someone subscribes to the state of extension 1234, Asterisk knows to give them the state of the SIP phone SIP/myphone. exten = 1234,hint,SIP/myphone Suppose that I want to write to a database the state of all my extensions, in order to display it in a web page. How could I do it using the hint mechanism? Thanks again, -- Dott. Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile type=
What are the values for type for chan_mobile? headset and phone ??? I get my Treo650 to pair. hcitool scan shows the device. hcitool con comes up empty. I go into Asterisk cli. mobile search shows the device (while I am waiting for a response, I see the phone showing a connection being set up). And I am shown that I have a device that is: NOT available and is a headset OOPS. So I think I need to force Asterisk to see this as a phone? Or is there something I need in a bluetooth config file? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel fallback
Hi list, My Asterisk v1.4 system now has two ISDN channels and two SIP channels. The idea is to make a dialplan that mostly uses the SIP channels for outgoing calls, but I'd like those to fall back automatically to ISDN if the SIP channels aren't available, possibly in combination with a warning issued to the caller before the call is actually placed. Is this possible with Asterisk? If so, how? Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console app
What does 'make menuselect' let you choose? Under #3, Channel Driveers, does chan_alsa have XXX through it so you can't select it? does chan_oss have XXX? This would indicate to you that the pieces of alsa or oss asterisk would need are not installed properly. Moj Gilberto Nunes wrote: Hi all I build an Asterisk, with asterisk 1.4.16.1 source. I have notice, that the console app don't appear on CLI... Is theres some options to turn on, when I compile asterisk? Thanks... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID blocking ...
On Jan 14, 2008 6:29 PM, Paul Hales [EMAIL PROTECTED] wrote: The 'setcallerpres' application is the one to use... Only works for PRI channels (maybe plain T1) channels via Zaptel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
On Tuesday 15 January 2008 16:03:16 Andrea Spadaccini wrote: Russell wrote: Andrea wrote: I have a small question: other than a phone (ie. SIP/something), what else can I use as app? Can I handle the change via some custom code? There are a number of things that can provide device state in Asterisk. That includes real devices such as SIP endpoints, or any other channel driver. However, it also includes things like monitoring the state of a space in parking, or the usage of a MeetMe conference. I have also written a small dialplan function which lets you create custom device states. A lot of people use this for things like having a light on the phone that reflects whether the agent is logged in or not. More information: http://asterisk.org/node/48325 http://asterisk.org/node/48360 Thanks a lot for the info, I already read the first article, and it's great to know that DEVSTATE can be used in 1.4. But my question was different, my poor english doesn't help me. :( In your article I read For example, when someone subscribes to the state of extension 1234, Asterisk knows to give them the state of the SIP phone SIP/myphone. exten = 1234,hint,SIP/myphone Suppose that I want to write to a database the state of all my extensions, in order to display it in a web page. How could I do it using the hint mechanism? Just create a module that subscribes to every single device and when the state changes, your callback will get an event with the device name that changed. You could then update your database with an SQL query (or whatever else you like). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'
Andrew Joakimsen wrote: Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' The problem is that 100101 is neither a valid IPv4 address nor a fully-qualified domain name. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channel fallback
Use the chanisavail to check that the SIP channels are clear, and set reasonable 'qualify=' settings for them PaulH On Tue, 2008-01-15 at 23:20 +0100, Jaap Winius wrote: Hi list, My Asterisk v1.4 system now has two ISDN channels and two SIP channels. The idea is to make a dialplan that mostly uses the SIP channels for outgoing calls, but I'd like those to fall back automatically to ISDN if the SIP channels aren't available, possibly in combination with a warning issued to the caller before the call is actually placed. Is this possible with Asterisk? If so, how? Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID blocking ...
On Tue, 2008-01-15 at 17:44 -0500, Andrew Joakimsen wrote: On Jan 14, 2008 6:29 PM, Paul Hales [EMAIL PROTECTED] wrote: The 'setcallerpres' application is the one to use... Only works for PRI channels (maybe plain T1) channels via Zaptel. Agreed entirely. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attended transfers manager or phone
Some phones have the auto-answer ability. So your phone could have two extensions, one for normal use and one for auto-answer use. Redirect or Originate, as you were, to the auto-answer extension on the phone. So the phone would already put itself offhook, and asterisk would continue and build up the other end of the bridge. Polycom soundpoint phones, for example, but many others have this ability. an example extension setup might be exten = 110,1,Dial(SIP/110) exten = #110,1,SipAddHeader(...whatever your phone needs to make it autoanswer) exten = #110,2,Dial(SIP/110) Don't know about phones that allow ip control of their state, though. Moj Christian Ejlertsen wrote: Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real simple answer to. I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Originate is part of the solution, but what really I want is the phone being taken off-hook and then being able to dial the number without having to answer the dial-back first. 1. One solution, though an ugly one, would be using Originate, but use a phone that has some sort tcp/ip interface that allows for taking the phone off-hook. 2. A Better solution would be using a phone that allows dialling and taking the phone off-hook on-hook etc. via some tcp/ip interface. 3. Yet another solution, though I do not favour this one since I really don't want to maintain the sip phone code, would be programming a soft sip phone with all the bells and whistles and adding the switchboard functionality to that (name searching, status email so on and so forth. In the end all I need is just a software or hardware phone, sip/iax, which can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps status requests. If such a phone exists that would do the trick, the rest is manageable via the Asterisk Manager console. I'm guessing some people have messed with this problem before so I hope that someone has some information about this kind of thing :) Thank you in advance Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile type=
Le mardi 15 janvier 2008, Robert Moskowitz a écrit : What are the values for type for chan_mobile? headset and phone ??? I think that if you let #type=headset it is by default a phone. It means that asterisk will try to use the mobile as a hand's free . If you set type=headset It means that asterisk will connect to the mobile as a bluetooth headset. I don't know what's the best in your case. For my mobile, w300i the rfcomm port number are 4 for handfreeprofile and 5 for headset profile. For the w300i, the handfree profile seems to be the best choice (yet no so good) port=4 ;handfreeprofile ; the rfcomm port number (from mobile search) ;port=5 ;headset profile chan_mobile did not told me the right rfcomm port number. I got the right number (at least I think) from kdebluetoothd that opens o konqueror window... with the mobile and gives all services. If I remember well, if you select services and copy them to a text editor you will get the rfcomm port. I get my Treo650 to pair. hcitool scan shows the device. hcitool con comes up empty. I go into Asterisk cli. mobile search shows the device (while I am waiting for a response, I see the phone showing a connection being set up). And I am shown that I have a device that is: NOT available and is a headset May not mean it won't work, I observed the same thing... OOPS. So I think I need to force Asterisk to see this as a phone? Or is there something I need in a bluetooth config file? You need a mobile.conf file Most usefull links for me for configuring chan_mobile were http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html http://www.saghul.net/blog/2007/08/29/howto-review-chan_mbile/ http://snapvoip.blogspot.com/2007/10/configuring-using-and-debugging.html more /etc/asterisk/mobile.conf [general] interval=10 ; Number of seconds between trying to connect to devices. [adapter] id=intuix address=00:11:67:2E:B4:FD ;forcemaster=yes ; attempt to force adapter into master mode. default is no. alignmentdetection=yes ; enable this if you sometimes get 'white noise' on asterisk side of the call [adapter] id=trendnet address=00:18:E7:2E:DC:AB ;forcemaster=yes ; attempt to force adapter into master mode. default is no. [W300i] address=00:16:B8:5F:C2:71 ; the address of the phone port=4 ;handfreeprofile ; the rfcomm port number (from mobile search) ;port=5 ;headset profile context=frommobile adapter=intuix ;adapter=trendnet dtmfskip=60 ;type=headset ; This is a headset, not a Phone ! group=1 ; this phone is in channel group 1 ;nocallsetup=yes; set this only if your phone reports that it supports call progress notification, but does not do it. Motorola L6 for example ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Call Recording
On Tue, 2008-01-15 at 12:13 -0600, Mike Hammett wrote: I'm a newb when it comes to patch. I have a combine_wave-0.3.orig and a combine_wave-0.3 directory. This is what I get: [EMAIL PROTECTED] ~]# patch combine_wave-0.3.patch can't find file to patch at input line 4 Perhaps you should have used the -p or --strip option? The text leading up to this was: -- |diff -Naur combine_wave-0.3.orig/combine_wave.c combine_wave-0.3/combine_wave.c |--- combine_wave-0.3.orig/combine_wave.c 2005-10-06 14:44:10.0 +0200 |+++ combine_wave-0.3/combine_wave.c2007-10-05 21:02:17.0 +0200 -- File to patch: Try this: $ tar -xvzf combine_wave-0.3.tgz $ patch -p1 combine_wave-0.3.patch $ cd combine_wave-0.3 $ make Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID blocking ...
On Tue, 2008-01-15 at 17:44 -0500, Andrew Joakimsen wrote: On Jan 14, 2008 6:29 PM, Paul Hales [EMAIL PROTECTED] wrote: The 'setcallerpres' application is the one to use... Only works for PRI channels (maybe plain T1) channels via Zaptel. Just to be more complete: SetCallerPres works with BRI chan_capi too. Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'
Well can you offer some explanation why T.38 faxing worked for months and then one day stopped working? Using both Linksys Audiocodes (yuck) ATA. The first second of the fax tone is heard and then the T.38 switchover is attempted and the call drops with said error. On Jan 15, 2008 6:25 PM, Mark Michelson [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' The problem is that 100101 is neither a valid IPv4 address nor a fully-qualified domain name. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help Unable to dial _99XXXXXXXX
Hi all This is rahul i am using asterisk 1.4.17 with degium TE120p card. I have configured the card but there is a problem coming Asterisk is dialing_98 series numbers but it is not dialing _99 showing CHANISUNAVAIL. Regards RAHUL ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'
16 jan 2008 kl. 04.43 skrev Andrew Joakimsen: Well can you offer some explanation why T.38 faxing worked for months and then one day stopped working? You are asking the wrong forum. Your device is clearly sending a bad SDP. Ask the vendor of that device. /O Using both Linksys Audiocodes (yuck) ATA. The first second of the fax tone is heard and then the T.38 switchover is attempted and the call drops with said error. On Jan 15, 2008 6:25 PM, Mark Michelson [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101' The problem is that 100101 is neither a valid IPv4 address nor a fully-qualified domain name. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * Olle E Johansson - [EMAIL PROTECTED] * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions
16 jan 2008 kl. 00.03 skrev Tilghman Lesher: On Tuesday 15 January 2008 16:03:16 Andrea Spadaccini wrote: Russell wrote: Andrea wrote: I have a small question: other than a phone (ie. SIP/something), what else can I use as app? Can I handle the change via some custom code? There are a number of things that can provide device state in Asterisk. That includes real devices such as SIP endpoints, or any other channel driver. However, it also includes things like monitoring the state of a space in parking, or the usage of a MeetMe conference. I have also written a small dialplan function which lets you create custom device states. A lot of people use this for things like having a light on the phone that reflects whether the agent is logged in or not. More information: http://asterisk.org/node/48325 http://asterisk.org/node/48360 Thanks a lot for the info, I already read the first article, and it's great to know that DEVSTATE can be used in 1.4. But my question was different, my poor english doesn't help me. :( In your article I read For example, when someone subscribes to the state of extension 1234, Asterisk knows to give them the state of the SIP phone SIP/ myphone. exten = 1234,hint,SIP/myphone Suppose that I want to write to a database the state of all my extensions, in order to display it in a web page. How could I do it using the hint mechanism? Just create a module that subscribes to every single device and when the state changes, your callback will get an event with the device name that changed. You could then update your database with an SQL query (or whatever else you like). The manager interface is our preferred connection to Asterisk from third-party modules. The AMI will report all device state changes, so you can create an app that updates your database based on this information. /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
