Re: [asterisk-users] PRI (Primary-NTT)

On 1/7/2013 9:22 AM, Dan Austin wrote: The last PRI I setup in Hong Kong was configured as a Primary-Net5, which maps to euroisdn in DAHDI. That was eight years ago, so things may have changed, but it is worth a try. You should also collect some Q.931 logs, as I have seen silly things like

[asterisk-users] PRI (Primary-NTT)

hi folks. i recently setup an Asterisk system in Hong Kong. their phone company told me that their T1 PRI switch type is Primary-NTT. however in chan_dahdi.conf there's no such option. i have it set to national. it worked fine for a while, but now suddenly stop working. in coming call just keep

Re: [asterisk-users] PRI got event HDLC Abort

2068 2068 Machine check polls ERR: 0 MIS: 0 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] PRI got event HDLC Abort

. On Fri, Nov 2, 2012 at 8:40 PM, Edwin Lam edwin@officegeneral.com mailto:edwin@officegeneral.com wrote: hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38

[asterisk-users] PRI got event HDLC Abort

pickupgroup=1 faxdetect=incoming context=defaultspan1 channel = 1-23 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Deleting OLD Voicemails

to renumber all the messages so that they are consecutive otherwise the voicemail application may give you grief. A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? yes -- Edwin Lam edwin@officegeneral.com

Re: [asterisk-users] Realtime Registration

]: res_config_pgsql.c:209 realtime_pgsql: Postgresql RealTime: Found 1 rows. [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Any suggestions??? Asterisk 1.4.42 -- Edwin Lam edwin@officegeneral.com Systems

Re: [asterisk-users] rasterisk not knowing config path?

in manuals. /usr/sbin/asterisk -r -s path_to_asterisk.ctl and you'll need both rw permission to that socket. (and minimum of x permission for the directory tree it resides in) -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http

[asterisk-users] skype connect early media

back tone for about 20 seconds and then become fast busy. is there any setting i'm unaware of when setting up sip w/ skype connect? any suggestions would be appreciated. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http

Re: [asterisk-users] How to use password file with AuthenticateApplication

. maybe you can upgrade yours -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth

Re: [asterisk-users] How to use password file with AuthenticateApplication

? e.g. echo -n 12345 | md5sum -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth

Re: [asterisk-users] Fax not detected by Asterisk

ideas what can I do to figure out why it does not detect the arriving faxes ? in your dialplan, did you Answer the call and Wait a few seconds for Asterisk to detect the fax tone before you do some other things? -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415

Re: [asterisk-users] Binding asterisk to two static IPs

when sending out packets on that subnet. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Multiple Asterisk Sessions on same machine

-addr:5060 this sets up port 6060 to forward to 5060. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Different callerid for different extensions

this out of my head, i haven't test it. but this should map all 100-399 extensions to DID 044578900-0444579199 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] DAHDI span timeing source

://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=co -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file

changes to the dialplan is required ? try something like: Monitor(wav,${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${EXTEN}) -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch

Re: [asterisk-users] Realtime and priority labels

on column values for sorting. i don't think having 'n' as the priority will sort the way you want. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] DTMF not being sent ( RFC2833 )

it if someone could give me some pointers on where to go next, what additionnal debugging steps I should perform. I would also really appreciate if someone could propose a solution. Please help! David Never give up, never surrender -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc

Re: [asterisk-users] Asterisk 1.8.3

everything work. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk 1.8.3

and can hear the early media from the other side. For Now 1.8.3..2 is very bad. agreed... -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Connecting Asterisk to Siemens Hipath 3750

this list's archive and found this: http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] fail-over server

servers for such purpose. however you can look into csync2 to sync all asterisk files between the 2 asterisk servers if you don't want extra hardware. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371

[asterisk-users] SILK codec

hi folks. i've been experimenting with SILK codec and meet with some success on incorporating it in pjsip (an open source sip client). now i'm trying to do the same thing on Asterisk. any documentations, pointers, etc i should look into? any help is appreciated. -- Edwin Lam edwin

Re: [asterisk-users] Configuring Softphone

and/or asterisk box behind NAT? -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth

Re: [asterisk-users] changing sip port

forwarding on OS level to achieve that. iptables -t nat -I PREROUTING -i you incoming interface -d your IP addr -p udp -m udp --dport 6080 -j DNAT --to-destination your ip addr:5060 p.s. the default SIP port is 5060. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph

Re: [asterisk-users] take input and store in variable

if no input for 7 sec. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] more condition check for gotoif

On 10/3/10 11:20 AM, Daniel Knoll wrote: Hello, is it possible to check more than one condition for GOTOIF in the dialplan? yes. check out asterisk expressions on wiki pages http://www.voip-info.org/wiki/view/Asterisk+Expressions -- Edwin Lam edwin@officegeneral.com Systems Engineer

Re: [asterisk-users] DAHDI fxstest?

fxstest? cd to tools directory then enter the command make menuselect select fxstest, save exit, then execute make. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

[asterisk-users] early media issue from phone co.

- PRI - phone co. i call the same cell# and if it's unavailable. the PRI return cause code 31 and hangup, asterisk will then send a SIP BYE to the sip phone and the channel will simply hangup. how do i get the message on the sip phone? -- Edwin Lam edwin@officegeneral.com Systems Engineer

Re: [asterisk-users] fax spandsp

is not in the library search path. Dumb, but true. The installation information page for spandsp, at http://www.soft-switch.org/installing-spandsp.html , warns about these issues. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370

Re: [asterisk-users] fax spandsp

i gave up on ReceiveFAX and uses iaxmodem/hylafax instead. Tommy Botten Jensen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Edwin Lam skrev: Klaus Darilion wrote: The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig

Re: [asterisk-users] fax spandsp

Klaus Darilion wrote: The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig and the the build options) did that. no effect. i've got exactly the same result. Edwin Lam wrote: Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam

[asterisk-users] fax spandsp

[Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op

Re: [asterisk-users] fax spandsp

Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up

Re: [asterisk-users] splitting sip.conf to two files

on audiocodes. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth

[asterisk-users] server unresponsive

a while the whole system will become unresponsive until i kill the asterisk process. i've checked our network switches/routers and connections. they all work fine without any packet lost. any suggestions? -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439

Re: [asterisk-users] server unresponsive

play with the qualify times. I did a bunch of this before I just gave up. I'm sure there is a better or proper way of handling this. I'm interested to hear it. Paul On Nov 20, 2009, at 3:41 PM, Edwin Lam wrote: hi folks. we've experienced some weird problems lately. we

[asterisk-users] transcoder card

. will the 121 person able to make calls? will it support more channels if i put 2 cards in the system? thanks. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Time of Day Routing

before 12:01. and your welcome-afternoon message will be played on after 12:01. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Here is Step by Step Example of Asterisk PBX System Install and configuration

editor. :-P forget about all these editors.. TECO is the Sh*t. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

[asterisk-users] IMAP voicemail storage.

i've been playing with 1.6 voicemail w/ IMAP storage. it seems to work fine. however once IMAP storage is enabled. everyone VM will use IMAP. is there a way to configure some users use IMAP and other users use traditional file base storage? -- Edwin Lam edwin@officegeneral.com Systems

Re: [asterisk-users] asterisk/E1

-root. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] a little regex help needed

). try: exten =s,n,Set(CALLERID(name)=${IF($[ ${REGEX(021245711.. 0${CALLERID(num)})} = 1] ? Office:${CALLERID(name)})}) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

is not the same as ethernet crossover cable. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation

Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1

after i upgraded it. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http

[asterisk-users] iax2_trunk_queue: Maximum trunk data space exceeded

. they never have this kind of problems. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Asterisk Data Calls

-- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] Out-Going Calleriid

= _9.,n,SetCallerPres(allow) before the Dial command in extensions.conf. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] iaxmodem + hylafax w/ DID routing

Doug Lytle wrote: Edwin Lam wrote: in FaxDispatch: FILETYPE=pdf case $CALLID4 in 1000) [EMAIL PROTECTED] 1001) [EMAIL PROTECTED] *) [EMAIL PROTECTED] esac This is also incomplete, actually there's no problems with the above script. it's just $CALLID4

[asterisk-users] iaxmodem + hylafax w/ DID routing

info. but in fact it's blank. any idea? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation

Re: [asterisk-users] Need some input for Quad T1 and channel banks.

wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? -- Edwin Lam [EMAIL

Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing

. it (Asterisk) segfault on rxfax when i enable faxdetect in zapata.conf. since then it disabled faxdetect and use nvfaxdetect function in dialplan, it works fine afterward. also it seems to works fine using regular 32bit kernel. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1

Re: [asterisk-users] Global Variables on Reload

reload, it resets to off again. I've tried setting the clearglobalvars=no as well as just commenting out that line, but no luck so far. Any ideas? we use MySQL db to store those global vars in our installation. i guess you can use and db to do that. -- Edwin Lam [EMAIL PROTECTED] Systems

[asterisk-users] Asterisk 1.4 w/ realtime static zapata

upgrading to 1.4? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] spandsp/tx_fax/rx_fax frustrations

. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] pulling my hair out over voicemail

check the file permission and make sure asterisk can read it. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth

[asterisk-users] Cisco 79xx XML services

hi guys. i'm writing some simple applications for the cisco 7970 services button. i read the asterisk wiki and it mention there's a CMXML_App_Guide.pdf file but there's nowhere can i find a link for it. does anybody know where can i find it? regards. -- Edwin Lam [EMAIL PROTECTED] Systems

Re: [asterisk-users] IBM x3400 w/ Digium TE220

support. yes. i did. the problem solved by getting the latest snapshot with subversion. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX

backupProxyPort/backupProxyPort emergencyProxy/emergencyProxy emergencyProxyPort/emergencyProxyPort outboundProxy/outboundProxy outboundProxyPort/outboundProxyPort registerWithProxytrue/registerWithProxy /sipProxies -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General

Re: [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX

qualify=yes or no. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation Provided by http://www.api

[asterisk-users] IBM x3400 w/ Digium TE220

=cn span=1,1,0,ccs,hdb3 span=2,0,0,esf,b8zs bchan=1-15 dchan=16 bchan=17-31 fxoks=32-55 any clues? p.s. the same setting works fine on HP Proliant server. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371

Re: [asterisk-users] Using Modems with Asterisk

at decent speeds (minimum of 28.8) that anyone knows of? If not, has anyone used a Digium FXS card for this? i've done: PRI - Asterisk - SIP to analog adapter - modem or fax and PRI - Asterisk - channel bank - modem or fax they both work fine. -- Edwin Lam [EMAIL PROTECTED] Systems

[asterisk-users] asterisk crashed

, libpri: 1.2.2 any clues would be appreciated? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation

Re: [asterisk-users] Caller ID Rewrite

/listinfo/asterisk-users -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com

Re: [asterisk-users] Question about TFTPD server

did you installed? make sure your /tftpboot directory and all the files inside is at least readable by everyone. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?

work. (maybe it's fixed in the newest firmware) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation

Re: [asterisk-users] Polycom SP4000 ftp problem

Carla Schroder wrote: On Monday 23 October 2006 17:38, Edwin Lam wrote: Re: [asterisk-users] Polycom SP4000 ftp problem From: Edwin Lam [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Carla Schroder wrote: Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't

Re: [asterisk-users] Polycom SP4000 ftp problem

Eric ManxPower Wieling wrote: rename bootrom.ld to something else like bootrom.ld-disabled. did that. it hung on sip.ld, rename sip.ld, it hung on phone1.cfg. seems like if the file is bigger than say 1k. it'll hang. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph

[asterisk-users] Polycom SP4000 ftp problem

but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the setting works fine w/ SP501 with bootrom 3.1.2/sip 1.6.3 -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op

Re: [asterisk-users] Polycom SP4000 ftp problem

and flexibility of ftp server. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com

[asterisk-users] zapata.conf pri

to me how to use these parementers or point me to some documentations on them? thanks. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

[asterisk-users] test tone

hi folks. does anybody know what's the phone number for SBC Nothern California's 102-type milliwatt test line? (specifically in 415 area code) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup

[Asterisk-Users] PRI in Shanghai China

= 1-15 channel = 17-31 any clues? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] Regarding cdr_manager.conf

- cdr_custom.conf Mysql - cdr_mysql.conf odbc - cdr_odbc.conf postgreSQL - cdr_pgsql.conf FreeTDS - cdr_tds.conf -- __ Edwin Lam [EMAIL PROTECTED] __ __ Systems Engineer, Office General, Inc. __ Ph: +1 415 439 4988 Fax: +1 415 283 3370

Re: [Asterisk-Users] more voicemail frustrations (was: realtime voicemail)

has anyone use the MailboxExists() function in dial plan? seems like no matter what i do. it'll just just execute the next proprity. :( -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op

[Asterisk-Users] linksys SPA-941

me a phone number to call, which appears to be a fax machine. that's when i gave up on those idiots. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

[Asterisk-Users] CDR problem

' and destination '', then the user at extension '' transfer the call to ''. the CDR always show '' as the src and '' as the dst. is there any way to change this behavior and have '' as the src? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988

Re: [Asterisk-Users] Dial command in extensions

= h,1,blah yeah. i figured that. but that would execute on everything in the context. Someone else suggested the g option on Dial. that might work better. i'll have to experiment on it. thanks. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1

[Asterisk-Users] Dial command in extensions

executed. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk

Re: [Asterisk-Users] cisco phones problems

in sip.conf: [1234] context=default type=friend host=dynamic username=1234 secret=test123 mailbox=1234 callerid=John Smith 1234 qualify=yes dtmfmode=rfc2833 any thoughts why this is happening? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283

[Asterisk-Users] cisco phones problems

the Grandstreams with Ciscos. the following is typical setting in sip.conf: [1234] context=default type=friend host=dynamic username=1234 secret=test123 mailbox=1234 callerid=John Smith 1234 qualify=yes dtmfmode=rfc2833 any thoughts why this is happening? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer

Re: [Asterisk-Users] cisco phones problems

102 (Non-critical Request) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation sponsored

Re: [Asterisk-Users] cisco phones problems

. we're not using Cisco switches but the server all the phone units are on the same vlan and i've set the QOS priority to highest on our switch for that vlan. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371

Re: [Asterisk-Users] Supervised transfer problem with BudgetTone

B rings ( to tell someone is in wait : A) So it seems to fail i believe the sequence should be: A call B B pushes flash button (A hears mp3) B calls C pressing send C answers B press transfer (B will hangup, A C is now connected) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office

[Asterisk-Users] Asterisk to PSTN

them together with some insane dialplan? or is there an easier way? any suggestions? comments? remarks? parameters? thx. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20

Re: [Asterisk-Users] Asterisk to PSTN

to E1 lines w/ DID seems out of the question. i guess the way to go is using channel banks to convert those to E1 then connect Asterisk that way. further research, how about using these: http://www.welltech.com.tw/product_e_03.htm will that work? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer

[Asterisk-Users] Sipura SPA-841

does anybody has experienece with Sipura SPA-841 phone unit? how's its sound quality especially speaker phone? i have several Grandstream phones and was getting fustrated about the quality and bugs of their firmware. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1

[Asterisk-Users] connecting Asterisk to NEC NEAX system

time if i dialed the 4 digits very fast it might get through. seems like there's a timming issue of the DTMF. what can i do to solve this? i looked through the docs for zaptel.conf zapata.conf and doesn't seems there's any parameters to control the DTMF timing. -- Edwin Lam [EMAIL PROTECTED] Systems

Re: [Asterisk-Users] connecting Asterisk to NEC NEAX system

have to set emdigitwait=1000 to get it to work reliably. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ Asterisk-Users