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Hello. There are so many sound files in /var/lib/asterisk/en. Is there an easy way to let me play back all of them one by one while I am watching CLI to see the current file name? Thanks for help. -- Jian Gao IT Technician SJ Geophysics Ltd. <http://www.sjgeophysics.com> jian.gao at sjgeophysics.com <mailto:jian.gao at sjgeophysics.com> Tel: (604)582-1100

...am sure the network part is working because I can use "dig" to find our their A record and IP address of the SIP proxy. Also, I do have srvlookup=yes in my sip.conf. But that doesn't help. Did I missing something here? -- Jian Gao IT Technician SJ Geophysics Ltd. <http://www.sjgeophysics.com> jian.gao at sjgeophysics.com <mailto:jian.gao at sjgeophysics.com> Tel: (604)582-1100

Hi, No matter I try, I can not register to Voipwise with Trixbox. It is always in "unregistered" state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com fromuser=username host=sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer

I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.....) [custom-inbound] exten => _556,1,answer exten => _556,n,playback(beep) exten => _557,1,answer exten => _557,n,playback(beep) exten => _558,1,answer exten => _558,n,playback(beep) exten =>

hi, I am quite new on using Samba and sorry maybe ask a silly question here. I set up simple Samba server on Fedora3 using the samba rpm package comes with fedora3( version 3.0.10-1.fc3). I use the SHARE security level to make things easier. Everything goes fine so far, except that for some windows user, some times, on browsing some directories, it takes extremely long time to display the

Hey all. My boss asked me to implement the following When DID 713xxxxxxx is dialed send an email to mmosier at xxx.com. with the time date and CID included in the email. I know how to code some but am looking for the best way to do this. Sorry I might have asked this a couple months back. I forgot. Mmosier Houston Respectfully Michael D Mosier Ftoc Certified -------------- next part

...-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110128/a65fcfd0/attachment-0001.htm> ------------------------------ Message: 5 Date: Fri, 28 Jan 2011 09:20:22 -0800 From: Jian Gao <jian.gao at sjgeophysics.com> Subject: Re: [asterisk-users] chan_sip bug? (Asterisk 1.4) To: asterisk-users at lists.digium.com Message-ID: <4D42FAD6.9030702 at sjgeophysics.com> Content-Type: text/plain; charset="us-ascii"; Format="flowed" Chad, You are right. tcpdump shows Asterisk sees 777...

Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via:

Centos 5.6 came out. Any one tried to update to the 5.6 yet? I am running Asterisk 1.8 and is there any risk to upgrade to Centos 5.6? -- *Jian* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110413/0a448985/attachment.htm>

Via: SIP/2.0/UDP 10.11.22.161:10000;branch=z9hG4bK-a860600e\x0d\x0a From: Jian Gao <sip:8181234567 at my.provider.com>;tag=7e9c4091bfc704bco0\x0d\x0a To: Jian Gao <sip:8181234567 at my.provider.com>\x0d\x0a Call-ID: daf96244-769f952c at 10.11.22.161\x0d\x0a CSeq: 48998 REGISTER\x0d\x0a Max-Forwards: 70\x0d\x0a Contact: Jian Gao <sip:8181234567 at

Hi, I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5) server. Everything seems fine but I just saw this WARNING shows up in the log every time I start the asterisk: /[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister/ And in later in the log

I am helping a friend on one of his sip trunk and couldn't find the way to resolve his problem. His asterisk's problem is like this: 0. When incoming call to one of his sip trunk, Asterisk reply with "488 Not acceptable here". So the call get dropped. 1. Recently upgraded Elastix with Asterisk 1.4.33 2. Was working fine before the upgrade 3. There are total 4 SIP trunks

Now in my asterisk config files, there are lines like: secret=some_password_in_plain_text Is it possible to hide these plain text password? -- *Jian * -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110214/7661ff8e/attachment.htm>