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Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at gmail.com -a ${FAXDEST}/${tempfax}.tif) I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by root. Any hint will be appreciated. El...

...re's a product that I can install on my debian-based server to extract CDRs (it'd be better if Excel's downloads are available), also it would be desirable if I can access additional table to update rows (e.g. sip for realtime) Please let me know what you know. Best Regards, Elder D. Arohuanca dCAP Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130320/c7d909cc/attachment.htm>

...ys/ Where 'ast-centos' is the result of 'uname -n' I've followed instructions from: http://goalbound.blogspot.com/2012/05/configure-asterisk-18110-on-centos-55.html and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial Any hint would be appreciated! Elder D. Arohuanca DCAP Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130206/22ad722e/attachment.htm>

...see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). This is happening with my PBX hosted on an external network and peers on my local network. It seems the SIP ACK is not being received properly. I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9 Elder D. Arohuanca Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130515/ea647eca/attachment.htm>

...es echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 signalling=fxo_ks context=local channel=>1-2 group=2 signalling=fxs_ks context=local channel=>3-4 *system.conf* fxoks=1,2 fxsks=3,4 echocanceller=mg2,1-4 loadzone = fr defaultzone = fr Thanks by your help Elder Arohuanca Lagos t. +51 1 994149553 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090222/bfe255e8/attachment.htm

...east 1 (I wish it could be zero but it isn't) I'd like to know if there's a way to do it. Setting the Queue() timeout to a little value is not the most suitable option. I'm using asterisk 1.4.21 but I don't know if there are some options available on release 1.8 Thanks, Elder Arohuanca Lagos t. 992728100 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110207/d1b9f2fd/attachment.htm>

...rt/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is not present" I apologize it is not a pure Asterisk question but I'm sure some of you can help me. Thanks in advance! Elder Arohuanca Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130412/dee24545/attachment.htm>

...l [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine odbc [Nov 25 11:09:26] NOTICE[24118] config.c: Registered Config Engine sqlite3 Any help would be welcome. My Linux distro is: Linux (my-ip-address) 3.11.6-x86-linode54 #1 SMP Wed Oct 23 15:22:49 EDT 2013 i686 GNU/Linux Elder D. Arohuanca Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131125/495dfa5f/attachment.html>

...addr=0.0.0.0 context=default language=es rtcachefriends=yes disallow=all allow=ulaw allow=alaw allow=gsm rtpholdtimeout=300 rtptimeout=300 dtmfmode=rfc2833 videosupport=yes progressinband=yes allowsubscribe=yes subscribecontext=extensiones notifyringing=yes notifyhold= yes limitonpeers= yes Daniel Arohuanca Lagos +51 1 994149553 Lima-Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081014/72edad82/attachment.htm

...;PRI RDSI - SPAN 1 group = 1 context = incoming-1 inmediate=no switchtype=euroisdn signalling=pri_cpe channel => 1-15,17-31 ;PRI RDSI - SPAN 2 group = 1 context = incoming-2 inmediate=no switchtype=euroisdn signalling=pri_cpe channel => 32-46,48-62 ............... Thanks in advance, Elder Arohuanca Lagos Phone: +51 1 991696900 Lima - Peru

...eer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an idea to solve this issue. Softswitch is using an implementation of RFC 3264 and the PBX being used is Elastix 2.3 with Asterisk 1.8.11.0 Thanks in advance Elder D. Arohuanca Lima - Peru *[1]* [Aug 12 19:21:05] VERBOSE[17115] app_dial.c: -- Called SIP/SIP-PROVIDER/965034648 *[2]* [Aug 12 19:21:14] WARNING[3477] chan_sip.c: Retransmission timeout reached on transmission 0e51f669152c660b3c97de1876d9e971@*PROVIDER-IP* for seqno 103 (Critical Request) -- See https://...

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121129/83c846a4/attachment.htm>

...h my current softphone and I`m trying to develop an IAX Client based one. Does anyone know how can I get help or useful resources about it? Specially with Conference function and management of incoming call events to launch an AGI at that time. I?ll be very gratefull for any help you have. Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080717/8e4d6212/attachment.htm

...npeer=yes *at sip.conf and every sip_buddie has *call-limit=2 *and *type=friend *but problem persists. My asterisk version is: 1.4.21.1 Calls are entering well now, but what when load increases high? Does anyone know what can I do to avoid this WARNINGS and future issues? Thanks again, Daniel Arohuanca +51 1 994149553 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081104/004d2822/attachment.htm

...ch=yes ;*groups* group=2 context=incoming immediate=no switchtype=euroisdn signalling=pri_cpe channel=32-46,48-62 #### *extensions.conf* exten => _9.,n,Dial(Zap/g2/${EXTEN:1}) Asterisk version is: 1.4.21.1 and I'm using a Digium TE420P card to connect to ISDN. Thanks in advance, Daniel Arohuanca +51 1 994149553 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081112/c2552f74/attachment.htm

Dear all, I wanna know what can I do to get the PBX's clock from -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090428/3218b8b0/attachment.htm

...e, channel SIP/AgentA-tag2 obtaining the values from SIP/AgentA-tag1, I could rename the callerID properly 2. If there is a way to get the values of the queue variables to relate agent with original caller, it could help too. Hope you can help me, I'm running Asterisk 1.4.21 Elder D. Arohuanca Lagos +51 1 945108658 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100319/bedca725/attachment.htm

...? I'm planning to do it to select the right provider to route further calls at least cost. In my country there are no public or accessible information on ported out numbers, so it is a way to discriminate what are the destination's Telco and build a database. Thanks in advance! Elder D. Arohuanca Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131223/3b089e71/attachment.html>

...t I have found this message on kern.log, messages and syslog.1 on /var/log/: May 27 09:48:32 pbx-thor-PE kernel: [334427.888524] asterisk[15384] general protection ip:482a13 sp:7f335b87c898 error:0 in asterisk[400000+221000] I am using Debian 7.5 64 bits with Asterisk 11.9.0 Thank you! Elder D. Arohuanca Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140528/21b05354/attachment.html>

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: