[asterisk-users] Insert name in SIP registry

Via: SIP/2.0/UDP 10.11.22.161:1;branch=z9hG4bK-a860600e\x0d\x0a From: Jian Gao sip:8181234...@my.provider.com;tag=7e9c4091bfc704bco0\x0d\x0a To: Jian Gao sip:8181234...@my.provider.com\x0d\x0a Call-ID: daf96244-769f952c@10.11.22.161\x0d\x0a CSeq: 48998 REGISTER\x0d\x0a Max-Forwards: 70\x0d

[asterisk-users] Safe to upgrade to Centos 5.6 now ???

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* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] Safe to upgrade to Centos 5.6 now ???

* On 11-04-13 01:04 PM, Shaun Ruffell wrote: On Wed, Apr 13, 2011 at 01:00:36PM -0700, Jian Gao wrote: 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? I'm not sure about Asterisk in general, but if you use

[asterisk-users] Error loading module 'res_fax_digium.so'

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':

Re: [asterisk-users] Hide the plain text password

How about encrypt the whole hard drive? If I built a server and give to other people, there is no easy way to stop them reset the root password or just mount my drive to read everything on it. But if build an encrypt OS then it will be secure. My question here are: 1Is this against Asterisk

[asterisk-users] Hide the plain text password

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 * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Hide the plain text password

I am building a server for a client. I want them to try out the new Google Voice feature using my GV account. But I don't want expose my GV's password. *Jian * On 11-02-14 01:46 PM, Kevin P. Fleming wrote: On 02/14/2011 03:36 PM, Jian Gao wrote: Now in my asterisk config files

Re: [asterisk-users] Gtalk/Jabber Issue

Port 5222 opened? Jian On 11-02-10 09:54 PM, William Stillwell wrote: I was getting unable to make channel.. So, this is what I am doing.. Service stop asterisk Purge modules Make clean Remove all traces of iskemel Recompile that. With , add needed entrée into ldconfig. Verify iksemel

Re: [asterisk-users] chan_sip bug? (Asterisk 1.4)

Chad, You are right. tcpdump shows Asterisk sees 777 when the packet arrived. It's truned out my router somehow modified the packet! I am using a Asus RT-N16 router with TomatoUSB firmware. There is a setting SIP Helper. I disabled this feature on the router then everything back to normal.

[asterisk-users] chan_sip bug? (Asterisk 1.4)

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:1778xxx@10.11.22.77:5060 SIP/2.0 Via:

Re: [asterisk-users] trixbox - sip trunk with voipwise

username=username Register string: username:p...@sip.voipwise.com mailto:username%3ap...@sip.voipwise.com Do you have any suggestions? Thank you. Mert -- Jian Gao IT Administrator SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582

[asterisk-users] 488 Not acceptable here

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

Re: [asterisk-users] Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny

can be your friend Joshua Stein has an great article on this topic: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/ -- Jian Gao -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] one for your filters

countries are not so straightforward I understand. Has anyone else tackled this problem? Thanks, j -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100

[asterisk-users] SIP SRV Registration problem

...@123.122.12.12, it works. I am using Asterisk 1.4 on Centos 5.4. I 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

Re: [asterisk-users] SIP SRV Registration problem

This is another error msg from CLI: [2010-05-17 11:48:50] WARNING[10957]: acl.c:400 ast_get_ip_or_srv: Unable to lookup 'proxy.provider.com' Jian Gao wrote: Hello, all, I have a Linksys 3102 from a VoIP provider. It use SRV record to register to the provider's SIP server. When I

Re: [asterisk-users] What is billsec in CDR?

that Asterisk will record the correct billsec? Or, is there a different approach? Place a ResetCDR() after your Playback() statement and before Dial(). Philipp ResetCDR() works! Thank you very much! -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian

[asterisk-users] What is billsec in CDR?

spent doing the playback in the billsec. CDR has two fields - duration and billsec. My understand is the billsec start count when a call is answered. Am I right? or Am I missing something here. -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com

Re: [asterisk-users] What is billsec in CDR?

ad...@3a.hu wrote: On 05-05-2010 18:00, Jian Gao wrote: In my system (Asterisk 1.4.30) I found that if I have some playback() or saydigit() before dial(), the billsec in CDR count all the time includes the playback time. For example, if I dial a number, listen the playback, then just

Re: [asterisk-users] Interesting email project.

} ${BODY} INLINE fi exit 0 -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

! http://track.sipfoundry.org/secure/attachment/16445/SPA_Provisioning_v3.pdf :) -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100

[asterisk-users] Playback all the sound files

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

Re: [asterisk-users] Playback all the sound files

than cooking up some asterisk way. -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] More efficient dial plan for a list of selective inbound numbers

of different prefixes, this list gets really big. Not desired. How can I have a more efficient dialplan? Thanks, Bruce -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100