[asterisk-users] Asterisk and cloud computing (amazon EC2 + S3)
Hey folks, I'm looking to potentially take some of my Asterisk servers and see how well they fare in a cloud computing environment such as Amazon EC2 + S3. I was curious to hear feedback from anyone who's willing to share their experience if they've already done the same. Have you had a positive experience and if not with Amazon, what other grid computing platform? Was it horrible and you'll never go back to it? Great ordeal of jitter/noise? Thanks a lot for your insight. :-) /sf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/IAX2 Provider with fallback dialing?
Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, /sf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
We're personally located in a small office based in Manhattan. Would need DIDs for the greater Manhattan area. But it sounds like Speakup is the type of service we're looking for that would cater to us domestically. On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- 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 aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP/IAX2 Provider with fallback dialing?
VoicePulse looks awesome, but they do not have the feature I need ... which is to be able to dial my mobile phone in the event my asterisk box or the Internet goes kablunk. On Thu, Jun 26, 2008 at 6:07 PM, Fred Posner [EMAIL PROTECTED] wrote: I think Voicepulse is out of NYC... not sure if they have failover though... but they have iax2 and sip. http://connect.voicepulse.com/ is their asterisk page. Fred Posner Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com FWD#: 902963 On Jun 26, 2008, at 5:56 PM, Steve Finkelstein wrote: We're personally located in a small office based in Manhattan. Would need DIDs for the greater Manhattan area. But it sounds like Speakup is the type of service we're looking for that would cater to us domestically. On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote: Hi all, I was curious if anyone can recommend a company that would work with small businesses, and capable of using a fallback number (mobile phone, home number etc) in the event SIP or IAX2 peering was to terminate because of some outage. This could be useful when you do not have a backup T1 PRI, etc. Thanks all, I dont know where you are, but here in .nl you can use Speakup. They route calls using IAX2 and/or SIP and in the case that wont work they will route it to another number you tell them (in my case, our support mobile number) -- 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 aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Local loopback vs SIP/IAX2
Hi all, Would anyone be able to point me in the right direction as far as the pros/cons of using a local loopback with a T1 provider, or just peering with a company using SIP/IAX2 or my small office asterisk setup? I've seen setups in both scenarios. The only potential pro of the T1 that I can think of is quality of voice and having an SLA with a provider. Otherwise, are the costs justified? Thanks for your input. /sf ___ -- 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] Looking for PSTN provider with unlimited inbound/outbound plan
Hi Senad, Did you happen to find out if it was indeed anywhere in the US48? Thanks! - sf On 1/2/08, Senad Jordanovic [EMAIL PROTECTED] wrote: Dovid B wrote: Senad, You can get unlimited as in FREE to any where in US48 or just local ? As far I know it is anywhere to US48. I will find out and get back to you. Senad - Original Message - From: Senad Jordanovic [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 31, 2007 3:10 PM Subject: Re: [asterisk-users] Looking for PSTN provider with unlimited inbound/outbound plan Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --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 ___ --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 . ___ --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] Looking for PSTN provider with unlimited inbound/outbound plan
Senad, Mind if I ask who that provider is? Thanks. Sent from my iPhone On Dec 31, 2007, at 8:10 AM, Senad Jordanovic [EMAIL PROTECTED] wrote: Justin Case wrote: Tell me when to stop laughing. Multiple channels and unlimited minutes ? No sane person will give that to you. Yap I agree... but but for about $900 per month one could get T1 (24 channels) unlimited in/out as far I seen last time our providers rates. Senad On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --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 ___ --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] Looking for PSTN provider with unlimited inbound/outbound plan
Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any potential leads. Happy holidays! - sf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cdr_adaptive_odbc and custom rdms fields
Hi folks,
I was recently made aware that the only way to currently set custom fields
in a relational database for CDR is via the experimental cdr_adaptive_odbc
drivers found here:
http://svncommunity.digium.com/view/tilghman/branches/1.4/cdr_adaptive_odbc.c?view=log
I had no problem compiling the driver, and copied the module to
/usr/lib/asterisk/modules on my box and setup
/etc/asterisk/cdr_adaptive_odbc.conf with the following:
[first]
connection=Asterisk-MySQL
table=cdr
My question is
a) How do I get asterisk to switch to the new module?
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: no
CDR registered backend: ODBC
CDR registered backend: csv
CDR registered backend: cdr-custom
CDR registered backend: cdr_manager
Looks like I'm still using the older stuff.
Ultimately my goal is to set a custom field in my CDR table that I will
populate with variables in my dialplan using the Set() application with
something such as:
; Set(CDR(dst_ext)=${DST_EXT})
Thanks again for your help.
- sf
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Re: [asterisk-users] cdr_adaptive_odbc and custom rdms fields
Hi Tilghman,
Any ideas on how to properly get this linked against my current Asterisk?
catalyst*CLI module load cdr_adaptive_odbc.so
[Dec 25 15:11:14] WARNING[22722]: loader.c:363 load_dynamic_module: Error
loading module 'cdr_adaptive_odbc.so':
/usr/lib/asterisk/modules/cdr_adaptive_odbc.so: undefined symbol: ast_verb
[Dec 25 15:11:14] WARNING[22722]: loader.c:649 load_resource: Module
'cdr_adaptive_odbc.so' could not be loaded.
Thanks and happy holidays!
On 12/25/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 25 December 2007 12:50:06 Steve Finkelstein wrote:
I was recently made aware that the only way to currently set custom
fields
in a relational database for CDR is via the experimental
cdr_adaptive_odbc
drivers found here:
http://svncommunity.digium.com/view/tilghman/branches/1.4/cdr_adaptive_odbc
.c?view=log
I wouldn't call them experimental; they simply arrived too late to be
included
directly in 1.4.
I had no problem compiling the driver, and copied the module to
/usr/lib/asterisk/modules on my box and setup
/etc/asterisk/cdr_adaptive_odbc.conf with the following:
[first]
connection=Asterisk-MySQL
table=cdr
My question is
a) How do I get asterisk to switch to the new module?
CLI module load cdr_adaptive_odbc.so
Looks like I'm still using the older stuff.
Ultimately my goal is to set a custom field in my CDR table that I will
populate with variables in my dialplan using the Set() application with
something such as:
; Set(CDR(dst_ext)=${DST_EXT})
--
Tilghman
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Re: [asterisk-users] cdr_adaptive_odbc and custom rdms fields
No worries!
Does something like this make sense for now?
//ast_verb(3, Found adaptive CDR table [EMAIL PROTECTED],
tableptr-table, tableptr-connection);
if(opt_verbose 2) {
ast_verbose(VERBOSE_PREFIX_3, Found adaptive CDR
table [EMAIL PROTECTED], tableptr-table, tableptr-connection);
}
Prolly my fault.
cc -I/usr/src/asterisk/include -I/usr/local/include
-DAST_MODULE=\cdr_adaptive_odbc\ -o cdr_adaptive_odbc.o -c
cdr_adaptive_odbc.c
cdr_adaptive_odbc.c: In function 'load_config':
cdr_adaptive_odbc.c:159: error: 'opt_verbose' undeclared (first use in this
function)
cdr_adaptive_odbc.c:159: error: (Each undeclared identifier is reported only
once
cdr_adaptive_odbc.c:159: error: for each function it appears in.)
cdr_adaptive_odbc.c: In function 'odbc_log':
cdr_adaptive_odbc.c:561: error: 'opt_verbose' undeclared (first use in this
function)
make: *** [cdr_adaptive_odbc.o] Error 1
On 12/25/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 25 December 2007 14:11:53 Steve Finkelstein wrote:
Any ideas on how to properly get this linked against my current
Asterisk?
catalyst*CLI module load cdr_adaptive_odbc.so
[Dec 25 15:11:14] WARNING[22722]: loader.c:363 load_dynamic_module:
Error
loading module 'cdr_adaptive_odbc.so':
/usr/lib/asterisk/modules/cdr_adaptive_odbc.so: undefined symbol:
ast_verb
[Dec 25 15:11:14] WARNING[22722]: loader.c:649 load_resource: Module
'cdr_adaptive_odbc.so' could not be loaded.
Crud, I know what's wrong, but unfortunately I can't fix it right now, as
there's a problem with the repository server (I can't connect).
In the meantime, you can replace in the source everywhere it says
ast_verb(3,
with
if (opt_verbose 2)
ast_verbose(VERBOSE_PREFIX_3
Sorry. We'll get this fixed as soon as someone with access to that server
logs on after vacation.
--
Tilghman
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Re: [asterisk-users] cdr_adaptive_odbc and custom rdms fields
Thanks, that did the trick! :-) On 12/25/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 25 December 2007 14:59:12 Steve Finkelstein wrote: No worries! Try updating from SVN now. I've made the change. -- 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 ___ --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] Reputable company for SIP/IAX2 trunking
Thanks for the reply. :-) Yeah. I need a service that's going to allow multiple channels eventually. Perhaps 1000 simultaneous calls through one number. I'm just doing my research now and it looks like I'm going to have to start getting multiple ds3s for this type of call center setup. On 12/16/07, Luki [EMAIL PROTECTED] wrote: Steve, if you want quality and reliability, then you need to get as close as you can to the actual big guys operating the equipment, such as Level3, GlobalCrossing, XO, CommPartners. But they won't be interested in doing business with you for just 1 DID and couple thousand minutes a month. So find yourself a good first-hand reseller of those big guys who is interested in doing business with you. There are many out there. We have been getting 90% of our west-coast DIDs from CommPartners directly, and over the last 3 years, I don't recall a single indecent when they let us down service. The actual VoIP service is excellent; billing and paperwork can be messy at times. Luki On Dec 15, 2007 4:25 PM, Steve Finkelstein [EMAIL PROTECTED] wrote: Hi all, There's a myriad of options these days and I haven't been keeping up to date with what's respectable any longer. I essentially need a provider that will provide me with one DID to start and let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on my end and have full control over the dial plan. This way I can branch out my DID into extensions and have it dial individual peers according to an extension. Looking for some feedback on what provider is quality these days. I don't mind paying an extra dollar or two. Thanks, - sf ___ --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] Reputable company for SIP/IAX2 trunking
Hi all, There's a myriad of options these days and I haven't been keeping up to date with what's respectable any longer. I essentially need a provider that will provide me with one DID to start and let me trunk over SIP or IAX2. I'd like to obviously trunk with Asterisk on my end and have full control over the dial plan. This way I can branch out my DID into extensions and have it dial individual peers according to an extension. Looking for some feedback on what provider is quality these days. I don't mind paying an extra dollar or two. Thanks, - sf ___ --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] GotoIf Dialplan inquiry
Hi all,
I have the following in my extensions.conf:
exten = s,4,GotoIf($[${CALLERID(number)} = 8585979857 |
8585970327]?15:5)
The numbers listed above are known spammer numbers. However, when I call
from any other CALLERID, it still directs me to s,15 which is the
Hangup() application. Here are logs from the asterisk CLI:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/8995SIP/31337|15|8995|IAX2/[EMAIL PROTECTED]/13476681546) in
new stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3, )
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3,
(8585970327)?15:5) in new stack
-- Goto (macro-forward,s,15)
-- Executing [EMAIL PROTECTED]:15] Hangup(IAX2/lime-3, ) in new stack
== Spawn extension (macro-forward, s, 15) exited non-zero on
'IAX2/lime-3' in macro 'forward'
== Spawn extension (macro-forward, s, 15) exited non-zero on 'IAX2/lime-3'
-- Hungup 'IAX2/lime-3'
Thank you for any assistance.
- sf
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[asterisk-users] Dial plan inquiry using GotoIf()
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone number
; arg2 = timeout
; arg3 = extension (voicemail)
; arg4 = mobile number
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Wait(1)
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
exten = s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
exten = s,6,AGI(didextlookup.agi|${CALLERID(number)})
exten = s,7,Set(CALLERID(number)=${didlookup})
exten = s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
exten = s,9,Set(CALLERID(number)=1${CALLERID(number)})
exten = s,10,Dial(${ARG1},${ARG2})
exten = s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
exten = s,12,Dial(${ARG4},${ARG2})
exten = s,13,Voicemail(u${ARG3})
exten = s,14,Playback(vm-goodbye)
exten = s,15,HangUp
exten = s,105,HangUp
As you can tell, exten = s,4,GotoIf($[${CALLERID(number)} =
15552221313]?15:5) is what I recently added.
Here's what I see in the CLI logs:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201) in new stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3, )
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3, 0?15:5) in
new stack
-- Goto (macro-forward,s,5)
It evaluates to false, hence goes to s,5. I keep dialing from that
particular number (the one in the example is clearly masked as a false
CID), and verified it's showing up as that number on callerID.
Also one last question. Say I need to add more numbers to block in the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Thank you kindly, everyone!
- sf
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Re: [asterisk-users] Dial plan inquiry using GotoIf()
Thanks for the help on this thread all.
It would make sense if I write an AGI and incorporate a DB backend to
check against numbers I want explicitly dropped. If anyone has such a
utility, I'd love to -not- reinvent the wheel. Otherwise, I'll go whip
it up and probably provide a web frontend for adding/removing numbers.
- sf
C F wrote:
It fails because the right function is ${CALLERID(num)}
On 5/30/07, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I'm looking for some rudimentary insight on GotoIf() which seems to be
failing on me in my dial plan. All I basically wish to do is block a
particular caller. Sounds easy enough, but my ternary operator/plan
currently is not properly being implemented. Can anyone spot where I'm
being a momo?
All extensions get forwarded to the following macro:
[macro-forward]
; arg1 = phone number
; arg2 = timeout
; arg3 = extension (voicemail)
; arg4 = mobile number
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,PrivacyManager
exten = s,3,Wait(1)
exten = s,4,GotoIf($[${CALLERID(number)} = 15552221313]?15:5)
exten = s,5,GotoIf($[${LEN(${CALLERID(number)})} = 4]?6:8)
exten = s,6,AGI(didextlookup.agi|${CALLERID(number)})
exten = s,7,Set(CALLERID(number)=${didlookup})
exten = s,8,GotoIf($[${LEN(${CALLERID(number)})} = 10]?9:10)
exten = s,9,Set(CALLERID(number)=1${CALLERID(number)})
exten = s,10,Dial(${ARG1},${ARG2})
exten = s,11,GotoIf($[${EXISTS(${ARG4})}]?11:12)
exten = s,12,Dial(${ARG4},${ARG2})
exten = s,13,Voicemail(u${ARG3})
exten = s,14,Playback(vm-goodbye)
exten = s,15,HangUp
exten = s,105,HangUp
As you can tell, exten = s,4,GotoIf($[${CALLERID(number)} =
15552221313]?15:5) is what I recently added.
Here's what I see in the CLI logs:
-- Executing [EMAIL PROTECTED]:1] Macro(IAX2/lime-3,
forward|SIP/5609|15|5609|IAX2/[EMAIL PROTECTED]/15164766201) in new stack
-- Executing [EMAIL PROTECTED]:1] Zapateller(IAX2/lime-3,
answer|nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] PrivacyManager(IAX2/lime-3, )
in new stack
-- CallerID Present: Skipping
-- Executing [EMAIL PROTECTED]:3] Wait(IAX2/lime-3, 1) in new
stack
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/lime-3, 0?15:5) in
new stack
-- Goto (macro-forward,s,5)
It evaluates to false, hence goes to s,5. I keep dialing from that
particular number (the one in the example is clearly masked as a false
CID), and verified it's showing up as that number on callerID.
Also one last question. Say I need to add more numbers to block in the
future, is there an easier way to do this than renumbering my entire
macro? Renumbering everything is just begging for a typo which can
effectively render my dial plan broken.
Thank you kindly, everyone!
- sf
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Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
This might be of some assistance: http://www.voip-info.org/wiki/view/vim+syntax+highlighting - sf Olivier wrote: Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards !DSPAM:1020,464b158e638175802679531! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1020,464b158e638175802679531! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.2 tanking CPU
Using a quad core 1.86GHz Xeon CPU here, running Asterisk 1.4.2. Noticed the following: Cpu(s): 4.3% us, 95.4% sy, 0.0% ni, 0.2% id, 0.0% wa, 0.0% hi, 0.0% si 30908 asterisk 18 0 188m 10m 5152 S 400 0.3 51051:13 asterisk Asterisk is eating up all the cores running the CPU at 400%. Is there something broken in 1.4.2 that needs to be addressed? Any suggestions? There's currently zero calls going on. - sf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk telemarketer torture sound files
What logic are you using to determine if the caller is indeed a telemarketer, anyway? Lacy Moore - Aspendora wrote: I like to forward them back to themselves, that is, the ones that give their phone number. Check nerdvittles.com. I think he had some kind of torture script setup, if I remember correctly. On 5/5/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Just forward them to 1-800-big-dick or some other 800 toll free phone sex line. They can't tell they've been forwarded. They'll figure it out eventually. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Jacob Muller Sent: Saturday, May 05, 2007 1:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk telemarketer torture sound files Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN). Anyone have a copy of these? -Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] telemarketer database ....next stages... Was asterisk telemarketer torture sound files
As good of an idea as that sounds, it would certainly need to be moderated in one form or another. It's just begging for abuse to allow any user to arbitrary insert numbers on the PSTN deeming them as blacklists. Sort of a spamhaus for telephony services .. Dave Bour wrote: How about a mysql sync - shared module that we could collectively poll periodically...to track exactly thisshare our telemarker numbers Would want 1. number, 2. label of who it is... 3. last hit on it... We'd each track our own...maybe an enhancement to the blacklist module... Use a peer to peer model...not unlike dundi...catch refreshes out of it Thoughts, commments, etc... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Jacob Muller Sent: Sunday, May 06, 2007 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk telemarketer torture sound files Manually, I'm just designating some people as telemarketers (using some MySQL + AGI to evaluate and drop the call) I would like to have a little fun. - Adam On May 6, 2007, at 12:06 PM, Steve Finkelstein wrote: What logic are you using to determine if the caller is indeed a telemarketer, anyway? Lacy Moore - Aspendora wrote: I like to forward them back to themselves, that is, the ones that give their phone number. Check nerdvittles.com. I think he had some kind of torture script setup, if I remember correctly. On 5/5/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Just forward them to 1-800-big-dick or some other 800 toll free phone sex line. They can't tell they've been forwarded. They'll figure it out eventually. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Jacob Muller Sent: Saturday, May 05, 2007 1:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk telemarketer torture sound files Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e- tools.com is NXDOMAIN). Anyone have a copy of these? -Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1020,463e1a17490845209328925! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using Playback() to play a random sound file
Unless there is some native rand() function available in Asterisk, I'd look into writing a simple AGI using Perl, PHP or Python to return back a random file to Playback(). More information here: http://www.voip-info.org/wiki-Asterisk+AGI HTH - sf Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? ~jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1020,46380d8a549331644115261! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 memory leak?
With all due respect, I believe you might be a bit paranoid. 10-11M is quite normal for the linux kernel to allocate for asterisk. It's not necessarily what the process is using, but that's just how memory management works within the kernel. What's 10-11M of RAM these days anyway? - sf Adam Moffett wrote: Is there a memory leak in asterisk 1.4? The other day with asterisk 1.4.0 I noticed that top was reporting a RES of 106 meg for the asterisk process. Restarting the process brought it down to more like 4 meg, but it grew over time to be 20+. So yesterday morning I upgraded to 1.4.4 in case this is something that had been addressed. Again I started with a RES of like 4meg or so, but this afternoon I'm up to 11megs: VIRT RES SHR SWAP CODE DATA 30932 11m 560818m 1012 17m Is this a real issue or do I have something else going on? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay in Dial()
All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in Dial()
It's amazing how simple some answer are.
Thank you kindly for your responses Edoardo and Luki. :-)
- sf
Edoardo Serra wrote:
Hi Steve,
put a timeout in the Dial command, if the call isn't answered it
returns after the timeout has expired
e.g.:
exten = _X.,1,Dial(SIP/${EXTEN}|15)
It waits 15 secs for the call to be answered
Look at http://www.voip-info.org/wiki-Asterisk+cmd+Dial for more
informations
Regards
Edoardo
Steve Finkelstein ha scritto:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile phone. Using something like a Wait() within a Dial()
would be ideal.
Any suggestions?
- sf
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[asterisk-users] Simple dial plan inquiry
Hi all,
This is a simple concept, however I'm not entirely comfortable with
available applications and functions available to me to make this happen.
I have a simple dialout macro such as the following:
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten = s,1,Set(CALLERID(number)=${ARG1})
exten = s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten = s,3,Set(ARG2=1${ARG2})
exten = s,4,Dial(${TRUNK}/${ARG2},,m)
exten = s,5,Congestion()exten = s,105,Busy()
This macro overrides one SIP endpoint which I use for personal usage and
do not wish to contain our default CID which is passed through arg1. Is
there anyway I can combine GotoIf/Goto to set it otherwise? I was
thinking in terms of pseudo code to do something similar to the following:
if ($arg1 = SIP/MyPersonal)
{
set caller ID to mypersonal
goto s,2
}
else
{
Set(CALLERID(number)=${ARG1}) ; leave as is
goto s,2 ; leave as is
}
Thanks for any insight.
- sf
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Re: [asterisk-users] Simple dial plan inquiry
Howdy Noah,
I just re-read my original inquiry and noticed my original purpose for
mailing the list was not simple to dig out of the message.
Ultimately, the dialout macro works fabulous. My issue is that I'd like
to be able to override one particular SIP endpoint with its own unique
callerID versus what is passed in $ARG1. So any exten that hits the
dialout macro will get set to the callerID in $ARG1. My one particular
SIP handset, for argument sake, SIP/123 .. should be set to CallerID = 234.
Does that clear up what I'm trying to accomplish some?
Thanks!
- sf
Noah Miller wrote:
Hi Steve -
[macro-dialout];
arg1 = callerid number;
arg2 = phone numberl
exten = s,1,Set(CALLERID(number)=${ARG1})
exten = s,2,GotoIf($[${LEN(${ARG2})} = 10]?3:4)
exten = s,3,Set(ARG2=1${ARG2})
exten = s,4,Dial(${TRUNK}/${ARG2},,m)
exten = s,5,Congestion()
exten = s,105,Busy()
This macro overrides one SIP endpoint which I use for personal usage and
do not wish to contain our default CID which is passed through arg1. Is
there anyway I can combine GotoIf/Goto to set it otherwise? I was
thinking in terms of pseudo code to do something similar to the
following:
Looks like it should work. Does it? Dialplan logic is fairly terse.
I don't think you'll be able to clean it up much more than that. If
you're looking for something that looks prettier, you could always use
AEL/AEL2. Of course, in the end AEL code will compile down to
Dialplan code.
- Noah
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Re: [asterisk-users] 100 users - voip lan security and qos ?
If you are using a cisco switch (2950, 3560, CE500, 4000, 6500, or 3750) then you will be able to setup the phone and have the computer daisy chained to it. I have a similar setup on mine. Here's how I configure my switch ports in order to achieve the desired effect: switchport access vlan 5 switchport voice vlan 6 auto qos voip cisco-phone This is assuming your data VLAN is configured as VLAN 5, and your VoIP VLAN is on VLAN 6. This will allow the phone to create a trunk port and facilitate both end nodes through one switch port. HTH - sf A_ Navone wrote: i have a customer that needs to plug the phones into the pc's using the pass-through rj45 available on most sip phones the question they are asking me is how to keep the data network separate from / secure from the voip network i understand they can set up vlans but i am hazy on a few details 1 since the phones are plugged into the pc's how will the phones be segmented into their own vlan ? 2 assuming the phone sends out a tos bit, how can we confirm that the customer's switch can read the tos bit and correctly prioritize it ? 3 to prioritize voip in the router (coming from the switch) we are looking at the wrtg54L and have found these 2 juicy websites http://openwrt.org and http://www.dd-wrt.com/dd-wrtv2/index.php has anyone downloaded and flashed the voip firmware ? does it give worthwhile advantages over the default firmware ? does the wrtg54L have any advantages over other routers ? any other advice to offer ? thank you so much in advance _ Exercise your brain! Try Flexicon. http://games.msn.com/en/flexicon/default.htm?icid=flexicon_hmemailtaglineapril07 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1020,4634f9c388295209328925! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold issue with asterisk 1.4.2
Interesting, that works David. I got the example directly out of the published VoIP Hacks book and followed instructions step by step. Either way, thanks much. :-) - sf Dave Miller wrote: Steve Finkelstein wrote on 4/28/07 12:21 AM: my musiconhold.conf: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 and finally in my extensions.conf: asterisk-1.4.2 # grep 100 /etc/asterisk/extensions.conf exten = 100,1,MusicOnHold(30) exten = 100,2,Hangup When I dial 100 however, I receive the following: [Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:947 local_ast_moh_start: No class: 30 The parameter to MusicOnHold is the class of music to play. You have no class named 30 just like the error says. :) You do have a class named default in the config snippet you pasted, so MusicOnHold(default) should work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music on Hold issue with asterisk 1.4.2
Hi all, I've compiled zaptel drivers and reconfigure asterisk afterwards from source --with-zaptel. Modules are loaded accordingly: asterisk-1.4.2 # lsmod |grep z Module Size Used by ztdummy 5472 0 zaptel194504 5 ztdummy crc_ccitt 3521 1 zaptel my musiconhold.conf: asterisk-1.4.2 # grep -v '^;' /etc/asterisk/musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 and finally in my extensions.conf: asterisk-1.4.2 # grep 100 /etc/asterisk/extensions.conf exten = 100,1,MusicOnHold(30) exten = 100,2,Hangup When I dial 100 however, I receive the following: -- Executing [EMAIL PROTECTED]:1] MusicOnHold(SIP/31337-007017f0, 30) in new stack [Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:947 local_ast_moh_start: No class: 30 [Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class '30') on channel SIP/31337-007017f0 == Spawn extension (internal, 100, 1) exited non-zero on 'SIP/31337-007017f0' Thanks for any input. - sf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cookbook
I'd definitely purchase this text, especially if Theodore Wallingford has any input on it. :-) - sf Doug Garstang wrote: What a cool idea! J. Oquendo wrote: http://etel.wiki.oreilly.com/wiki/index.php/Main_Page ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1020,4631203a316575802679531! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7960G + Asterisk auto attendant
All, I'm trying to hear the asterisk's auto attendant in its default configuration. According to VoIP Hacks in Chapter 4, I found the following excerpt after successfully configuring my SIP IP Phone (Cisco 7960G): In its default configuration, Asterisk has an auto-attendant that can route calls. To try it out, take the IP phone off the hook and dial 2. Then dial the BudgeTone's Send button. You will hear a friendly voice saying, Asterisk is an open source, fully featured PBX and IVR platform…. However, when I dial '2' on the phone, I just get a busy signal. Through the CLI it looks to have the demo available: vitamin-nybw*CLI console dial 2 [Apr 24 12:34:35] WARNING[8070]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory -- Executing [EMAIL PROTECTED]:1] BackGround(OSS/dsp, demo-moreinfo) in new stack Console call has been answered -- OSS/dsp Playing 'demo-moreinfo' (language 'en') [Apr 24 12:34:36] WARNING[8071]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory Any idea why I can't hear the asterisk default demo when dialing 2? - sf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] C7960 TFTP [Slightly off-topic]
Hi all, This is slightly off-topic, but I was hoping to be able to receive some insight as I'm sure plenty of experts with c7960's exist on this mailing list. I'm attempting to upgrade from SIP 8.3 - 8.6 on a C7960G that I inherited. I have my TFTP setup and unfiltered. The phone is doing TFTP over the internet as I'm telecommuting today, but I've placed it on the dmz to avoid any firewall headaches. Here's what a packet capture looks like: tcpdump -vv -Xnni eth1 -s 1000 port 69 tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size 1000 bytes 16:30:25.649429 IP (tos 0x0, ttl 51, id 1612, offset 0, flags [none], proto: UDP (17), length: 56) 1.2.3.4.50770 64.90.184.96.69: [no cksum] 28 RRQ SIP00187330C526.cnf octet 0x: 4500 0038 064c 3311 b2a8 44c1 9145 E..8.L..3...D..E 0x0010: 405a b860 c652 0045 0024 0001 5349 @Z.`.R.E.$SI 0x0020: 5030 3031 3837 3043 3532 362e 636e P00187330C526.cn 0x0030: 6600 6f63 7465 7400 f.octet. 16:30:41.667380 IP (tos 0x0, ttl 51, id 1617, offset 0, flags [none], proto: UDP (17), length: 56) 1.2.3.4.50771 1.2.3.5.69: [no cksum] 28 RRQ SIP00187330C526.cnf octet 0x: 4500 0038 0651 3311 b2a3 44c1 9145 E..8.Q..3...D..E 0x0010: 405a b860 c653 0045 0024 0001 5349 @Z.`.S.E.$SI 0x0020: 5030 3031 3837 3043 3532 362e 636e P00187330C526.cn 0x0030: 6600 6f63 7465 7400 f.octet. 16:31:15.596217 IP (tos 0x0, ttl 51, id 1606, offset 0, flags [none], proto: UDP (17), length: 51) 1.2.3.4.50757 1.2.3.5.69: [no cksum] 23 RRQ SIPDefault.cnf octet 0x: 4500 0033 0646 3311 b2b3 44c1 9145 E..3.F..3...D..E 0x0010: 405a b860 c645 0045 001f 0001 5349 @Z.`.E.E..SI 0x0020: 5044 6566 6175 6c74 2e63 6e66 006f 6374 PDefault.cnf.oct 0x0030: 6574 00 et. 16:31:31.621286 IP (tos 0x0, ttl 51, id 1611, offset 0, flags [none], proto: UDP (17), length: 58) 1.2.3.4.50758 1.2.3.5.69: [no cksum] 30 RRQ ./SIP00187330C526.cnf octet 0x: 4500 003a 064b 3311 b2a7 44c1 9145 E..:.K..3...D..E 0x0010: 405a b860 c646 0045 0026 0001 2e2f @Z.`.F.E../ 0x0020: 5349 5030 3031 3837 3043 3532 362e SIP00187330C526. 0x0030: 636e 6600 6f63 7465 7400 cnf.octet. 16:31:47.652531 IP (tos 0x0, ttl 51, id 1617, offset 0, flags [none], proto: UDP (17), length: 58) 1.2.3.4.50759 1.2.3.5.69: [no cksum] 30 RRQ ./SIP00187330C526.cnf octet 0x: 4500 003a 0651 3311 b2a1 44c1 9145 E..:.Q..3...D..E 0x0010: 405a b860 c647 0045 0026 0001 2e2f @Z.`.G.E../ 0x0020: 5349 5030 3031 3837 3043 3532 362e SIP00187330C526. 0x0030: 636e 6600 6f63 7465 7400 cnf.octet. 16:32:03.679382 IP (tos 0x0, ttl 51, id 1623, offset 0, flags [none], proto: UDP (17), length: 58) 1.2.3.4.50760 1.2.3.5: [no cksum] 30 RRQ ./SIP00187330C526.cnf octet 0x: 4500 003a 0657 3311 b29b 44c1 9145 E..:.W..3...D..E 0x0010: 405a b860 c648 0045 0026 0001 2e2f @Z.`.H.E../ 0x0020: 5349 5030 3031 3837 3043 3532 362e SIP00187330C526. 0x0030: 636e 6600 6f63 7465 7400 cnf.octet. --- The phone also displays TFTP SIP00187330C526.cnf on its LCD, however it does not appear to be retrieving binaries from the tftpserver. tftproot # cat OS79XX.TXT P003-08-6-00 -rwxr-xr-x 1 root root 129824 Dec 12 16:54 P003-08-6-00.bin -rwxr-xr-x 1 root root 130228 Dec 12 17:21 P003-08-6-00.sbn -rwxr-xr-x 1 root root459 Dec 12 17:40 P0S3-08-6-00.loads -rwxr-xr-x 1 root root 753560 Dec 12 17:20 P0S3-08-6-00.sb2 -rw-r--r-- 1 root root 681556 Jan 10 14:55 P0S3-08-6-00.zip -rw-rw-rw- 1 root root779 Apr 20 14:21 SIP00187330C526.cnf -rw-rw-rw- 1 root root 4658 Apr 20 16:29 SIPDefault.cnf -rw-rw-rw- 1 nobody nobody 11675652 Mar 28 16:18 c2600-entbase-mz.123-22.bin -rw-rw-r-- 1 nobody nobody 7735532 Jan 16 15:54 c2600-i-mz.123-21.bin -rw-rw-rw- 1 nobody nobody 11100664 Jan 16 16:24 c2600-ik9s-mz.122-27.bin -rw-rw-r-- 1 nobody nobody 8450865 Feb 27 23:35 c3560-advipservicesk9-mz.122-35.SE1.bin -rw-rw-rw- 1 nobody nobody 2726 Jan 16 16:23 cmeinternetlink-confg -rw-r--r-- 1 root root223 Apr 20 14:13 dialplan.xml -rw-rw-rw- 1 nobody nobody 1413 Jan 16 16:10 helfant-confg -rw-r--r-- 1 root root779 Apr 20 14:11 xmlDefault.CNF.XML Thanks for any suggestions all, - sf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering
Hi all, I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my existing Cisco 7960G handset(s). I've tried multiple installs of asterisk 1.4.2 with multiple handsets and SIP will not authorize my phone. I'll include some verbose log messages below to show a VALID registration and one where I'm having difficulty registering the phone. Thanks to anyone who can lend a helping hand with this matter or offer any insight on how to further debug. I've gone as far as packet capture and cannot understand why using the same configs will not allow registration of these handsets. - sf -- Working excerpt: REGISTER sip:10.2.7.2 SIP/2.0 Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK1029a1dd From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 104 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----0017e0134094;+u.sip!model.ccm.cisco.com=7 Content-Length: 0 Expires: 120 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK1029a1dd;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK1029a1dd;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED];tag=as010f0581 Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3f28f962 Content-Length: 0 REGISTER sip:10.2.7.2 SIP/2.0 Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK2596e8d7 From: sip:[EMAIL PROTECTED];tag=0017e0134094000e1fcda391-7d3d3796 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 104 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----0017e0134094;+u.sip!model.ccm.cisco.com=7 Content-Length: 0 Expires: 120 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK2596e8d7;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000e1fcda391-7d3d3796 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK2596e8d7;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000e1fcda391-7d3d3796 To: sip:[EMAIL PROTECTED];tag=as0a5554ff Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=5f5d830d Content-Length: 0 REGISTER sip:10.2.7.2 SIP/2.0 Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK5153628e From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 105 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;+sip.instance=urn:uuid:----0017e0134094;+u.sip!model.ccm.cisco.com=7 Authorization: Digest username=6096,realm=asterisk,uri=sip:10.2.7.2,response=6bec57e7aaedd046469fab89b39c024a,nonce=3f28f962,algorithm=MD5 Content-Length: 0 Expires: 120 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK5153628e;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK5153628e;received=10.2.7.254 From: sip:[EMAIL PROTECTED];tag=0017e0134094000d045a4ac2-6d217bab
Re: [asterisk-users] Asterisk 1.4.2 + Cisco 7960G not registering
Doug, Were you also having issues specifically related to SIP authorization, same as we're experiencing? I noticed other folks on the list aren't having any issues either. Did it just work out of the box? Thanks .. - sf Doug Lytle wrote: Steve Finkelstein wrote: Hi all, I've recently upgraded to Asterisk 1.4.2, coming from 1.2.14. Using my existing Cisco 7960G handset(s). I've tried multiple installs of asterisk 1.4.2 with multiple handsets and SIP will not authorize my User-Agent: Cisco-CP7960G/8.0 I was only able to get a stable setup after I moved my 7940 back to SIP version 7.5 Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
