Re: [asterisk-users] Manager API ! (System) command
On Wednesday 10 October 2007 07:04:02 robert home wrote: I need to issue some system commands via the Asterisk manager API. From the CLI the ! (system command) works fine, but when connected via the manager API it fails. Does anyone know why, or of a work around? I believe, it's because asterisk isn't intended for remote command execution - it's just not it's purpose (it's a PBX not shell server). I suppose the code of handling ! is in client part of asterisk CLI, not server. There are other far much superior and faster ways how to do that. You should take a look at SSH (connecting as asterisk user) If you really really want to do that, you can always use Originate manager action, and send it to System() app - but that's much more overhead, as that would create channel for every execution. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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] Wifi - Managed access points...
Luis, I strongly recommend that you test the setup before deployment. I have done a lot of tests with WiFi VoIP, handover, security, and though I don't have experience with the hardware you mention, I know WiFi VoIP is very brittle, especially in combination with WPA and handover. Battery life is a very important concern. And maybe you can report your findings to this list? Luis Antonio Prata Barbosa schreef: Hi, I'm working on a Wifi VoIP project specification. It will have almost 8 APs and 20-30 wifi phones. And after some research, I still having some questions ... 1) Are Managed Access Points (and switch controllers) really important to implement good wifi woip (w/ low latency and acceptable handover time) ? 2) What is the difference between (3com WX1200 + 3com AP 3750) and (DES-1228P + DWL-3140AP) ??? 3) 3com says their AP implements WMM ... and DLink says they priorize VoIP traffic based on VLAN ... are those methods the same ? No, they are not. Ron Arts Thank you, Luis A P Barbosa ___ --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] Why Asterisk doesn't accept sip302 redirect?
Vitaly,
Can you provide details of what is going on in the packet capture exactly?
What is the Contact: URI that the peer provides in the 302 Moved response?
What does Asterisk do subsequently?
Cheers,
-- Alex
On Wed, 10 Oct 2007, Vitaly wrote:
My asterisk should follow 302 redirect which it
receives from other sip server(10.10.10.10). By
running network sniffer I see, that asterisk receives
302 answer, but doesn't follow it.
My config is:
sip.conf:
...
[out4]
type=peer
host=10.10.10.10
canreinvite=no
promiscredir=yes
insecure=very
disallow=all
allow=g729
allow=g723
...
extensions.conf:
[to-sip]
exten = _0011X., 1, Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _0011X., 2, Hangup()
Any ideas?
Vitaly
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[asterisk-users] Why Asterisk doesn't accept sip302 redirect?
My asterisk should follow 302 redirect which it
receives from other sip server(10.10.10.10). By
running network sniffer I see, that asterisk receives
302 answer, but doesn't follow it.
My config is:
sip.conf:
...
[out4]
type=peer
host=10.10.10.10
canreinvite=no
promiscredir=yes
insecure=very
disallow=all
allow=g729
allow=g723
...
extensions.conf:
[to-sip]
exten = _0011X., 1, Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _0011X., 2, Hangup()
Any ideas?
Vitaly
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Re: [asterisk-users] Why Asterisk doesn't accept sip302 redirect?
Thanks for your answer, see details below:
U 10.10.10.10.67:5060 - 10.10.10.107:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0..v:
SIP/2.0/UDP
10.10.10.67:5060;branch=z9hG4bK0264a8da;rport..f:
2519494
sip:[EMAIL PROTECTED];tag=as1d5e5664..t:
sip:[EMAIL PROTECTED]..m:
sip:[EMAIL PROTECTED]..i: 503f1f3a
[EMAIL PROTECTED]: 102
INVITE..User-Agent: Asterisk PBX..Max-Forwards:
70..Date: Wed, 10 Oct 2
007 10:01:31 GMT..Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..c:
application/sdp..l: 259v=0
..o=root 2423 2423 IN IP4
10.10.10.67..s=session..c=IN IP4 10.10.10.67..t=0
0..m=audio 17250 RTP/AVP 18 4 101..a=rtpmap
:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:4
G723/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-16..a=silence
Supp:off - - - -..
#
U 10.10.10.107:5060 - 10.10.10.67:5060
SIP/2.0 302 Redirect..Contact:
sip:[EMAIL PROTECTED]:11060..v: SIP/2.0/UDP
10.10.10.67:5060;branch=z9hG4bK0264
a8da;rport..CSeq: 102 INVITE..Content-Length: 0
Master.csv:
,2519494,001112345678,to-sip,2519494,SIP/10.10.10.66-09e0a8b0,SIP/out4-09e15578,Dial,SIP/12345678
@out4,2007-10-10 15:01:31,,2007-10-10
15:02:01,30,0,NO ANSWER,DOCUMENTATION
--- Alex Balashov [EMAIL PROTECTED] wrote:
Vitaly,
Can you provide details of what is going on in the
packet capture exactly?
What is the Contact: URI that the peer provides in
the 302 Moved response?
What does Asterisk do subsequently?
Cheers,
-- Alex
On Wed, 10 Oct 2007, Vitaly wrote:
My asterisk should follow 302 redirect which it
receives from other sip server(10.10.10.10). By
running network sniffer I see, that asterisk
receives
302 answer, but doesn't follow it.
My config is:
sip.conf:
...
[out4]
type=peer
host=10.10.10.10
canreinvite=no
promiscredir=yes
insecure=very
disallow=all
allow=g729
allow=g723
...
extensions.conf:
[to-sip]
exten = _0011X., 1, Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _0011X., 2, Hangup()
Any ideas?
Vitaly
Be a better Heartthrob. Get better relationship
answers from someone who knows. Yahoo! Answers -
Check it out.
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Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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Re: [asterisk-users] get egress SIP call Id
Hello Steve,
I think Ray was talking more like the following setup (do correct me if
I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case, the INVITE SIP callId received by Asterisk from User A is
different to that sent in the INVITE to User B.
I can get User A's callId using ${SIPCALLID}. How about accessing SIP
callid of the INVITE sent to User B??
Typical need for this, is to store both the callIds to store in the CDRs
for debugging purposes(w.r.t. the service provider, et al).
cheerz
- Ben.
Steve Totaro wrote:
You can capture the sipcallid from the manager output. The cool part is
that the sipcallid is the same on both sides of a call. So,
AsteriskA---SIP (sipcallid) AsteriskB SIP (Same sipcallid as
AsteriskA for that call.
It is really easy to capture it from the manager.
Thanks,
Steve
Ray Chen wrote:
Hi Philipp,
Thank you for your response to my question. I am working on a
project which uses Asterisk as the voice engine. I need to
get the ingress and egress sip call id for a call to write call CDR.
(Asterisk CDR does not meet our customer requirments). If there is
no any easy way to get it I might need to create a seperate
process/thread to query manager interface as you mentioned. Thanks you,
Ray
Ray Chen wrote:
Hi, Does anybody know how to get the SIP call ID of a Dial
command?
There's no easy way to do it. What's your
intention? There are several events on the
manager interface.
Regards,
Philipp Kempgen
--
T
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EMAIL DISCLAIMER : This email and any files transmitted with it are
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prohibited. If you receive this transmission in error, please notify the sender
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shall be understood to be neither given nor endorsed by Mascon. Any information
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Re: [asterisk-users] get egress SIP call Id
Also, how do you acces the second SIP call ID from the dialplan? Any
simple way to do this?
Benjamin Jacob wrote:
Hello Steve,
I think Ray was talking more like the following setup (do correct me
if I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case, the INVITE SIP callId received by Asterisk from User A
is different to that sent in the INVITE to User B.
I can get User A's callId using ${SIPCALLID}. How about accessing SIP
callid of the INVITE sent to User B??
Typical need for this, is to store both the callIds to store in the
CDRs for debugging purposes(w.r.t. the service provider, et al).
cheerz
- Ben.
Steve Totaro wrote:
You can capture the sipcallid from the manager output. The cool part
is that the sipcallid is the same on both sides of a call. So,
AsteriskA---SIP (sipcallid) AsteriskB SIP (Same sipcallid as
AsteriskA for that call.
It is really easy to capture it from the manager.
Thanks,
Steve
Ray Chen wrote:
Hi Philipp,
Thank you for your response to my question. I am working on a
project which uses Asterisk as the voice engine. I need to
get the ingress and egress sip call id for a call to write call CDR.
(Asterisk CDR does not meet our customer requirments). If there is
no any easy way to get it I might need to create a seperate
process/thread to query manager interface as you mentioned.
Thanks you,
Ray
Ray Chen wrote:
Hi, Does anybody know how to get the SIP call ID of a Dial
command?
There's no easy way to do it. What's your
intention? There are several events on the
manager interface.
Regards,
Philipp Kempgen
--
T
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prohibited. If you receive this transmission in error, please notify the sender
by reply email and then destroy the message. Opinions, conclusions and other
information in this message that do not relate to official business of Mascon
shall be understood to be neither given nor endorsed by Mascon. Any information
contained in this email, when addressed to Mascon clients is subject to the
terms and conditions in governing client contract.
Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we
can not guarantee that any email or attachment is free from computer viruses
and you are strongly advised to undertake your own anti-virus precautions.
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Re: [asterisk-users] inbound call voip providers
http://www.didww.com/ will provide numbers. They even have a neat test thing on their website where you can set it up to work with your box. I haven't subscribed to them, but they seem ok. Here is the voip-info link with the full DID provider list. http://www.voip-info.org/wiki/view/DID+Service+Providers Todd On Oct 9, 2007, at 5:05 PM, srgqwerty wrote: Rafael: Thanks for your reply. I browsed http://www.fonetglobal.com but it seems to have local numering only in America. We need this service but in Europe. Do you have this service in Europe? The thing that we need is pretty simple. When the user calls a normal PSTN phone# from his normal PSTN telephone the provider stablishes a SIP session over IP to our asterisk box. Regards On Monday 08 October 2007 23:08, Rafael Canchola wrote: http://www.fonetglobal.com ___ --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] maximum retries exceeded on transmission Warnings
Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission SIPcallId for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids mentioned in the warnings are of the inbound leg. I've scoured the net, but haven't got anything conclusive. Have found responses ranging from firewall issues, no reception of ACKs, to bugs in some versions of Asterisk. I am using Asterisk 1.4.4, all SIP calls, with PSTN termination provided by my service provider. Have no firewalls or iptables set on my server. The calls did not seem to work even across a restart of asterisk. Interestingly, the calls to and from the very same numbers worked later on the next day. Anyone faced similar problems and was able to get the root of it? Or is it a bug? cheerz - Ben EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Loading Screen in Asterisk Gui
Hello, When I click on User menu, I get loading screen status. It runs indefinitely without showing me the user list and the user admin menu. Any thoughts ? Thanks, Sanjoy. Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/ ___ --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] Manager API ! (System) command
Yes - use the manager API to do an Originate by setting variable $CMD to the shell code you want to execute, and then call a piece of dialplan where the shellout will be executed through the System( $CMD ) command. Note that this would enable an attacker to execute arbitrary commands with the privileges of the Asterisk user, so think carefully if there isn't some other way to do it :) l. In data Wed, 10 Oct 2007 06:04:02 +0200, robert home [EMAIL PROTECTED] ha scritto: I need to issue some system commands via the Asterisk manager API. From the CLI the ! (system command) works fine, but when connected via the manager API it fails. Does anyone know why, or of a work around? Thanks Robert -- Home of QueueMetrics - http://queuemetrics.com ___ --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] Sine Dialer, GNU dialer, VICIDial and others slightly OT?
Hello John, we have a number of customers using each of the solutions you mention and they all seem to be working correctly. Unless you need a very unusual or extremely large setup, my suggestion is to go for the one that better fits your problem space / usage needs. I hope this helps l. On Tue, 09 Oct 2007 00:04:08 +0200, John Millican [EMAIL PROTECTED] wrote: Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of integration with asterisk, stability, and such. Thank You for any help JohnM -- Loway Research - Home of QueueMetrics http://queuemetrics.com ___ --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] transferring callerid ?
I'm expanding our tiny asterisk setup with a couple of external SIP phones, and I've just noticed the issue of the callerid not being displayed on an attended transfer. This bug seems to deal with it: http://bugs.digium.com/print_bug_page.php?bug_id=8824 I'm surprised that this hasn't been dealt with a long time ago - is there perhaps a work-around that I'm not aware of? /Per Jessen, Zürich ___ --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] Sangoma vs Digium: (was Re: ping too)
Just as a follow up on this thread, I decided to go for the Digium 412P quad port card. Thanks to everyone who commented, positively and negatively - it helped provide a balanced view in the end. Julian. Matt Florell wrote: On 10/6/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Julian Lyndon-Smith wrote: Julian Lyndon-Smith wrote: Nothing from me is posting to the list either. heh. Thought that this trick would work: it did for Doug. I've been trying to send the email below for 3 days now ! I know this is probably going to ignite the flames again .. I have looked at the recent threads regarding these two manufacturers, but there didn't seem to be much *technical* differences between the 2, it was rather more subjective - some people say Sangoma is better, some say Digium. And quite a lot of you're spreading FUD No, you are etc. I wanted to know if anyone has any specific comparisons or suggestions on the Sangoma A104D and the Digium TE406/411 cards ? I didn't want to start a flame war. I honestly just wanted a simple yes or no to the question Is a 406/411 technically comparable to the a104D. I think you mean the TE407/412 cards, the 406/411 series (using the OKI chipset instead of the Octasic) were discontinued by Digium. And while the Sangoma line uses a104 as a base for all variations of their quad-port T1/E1 cards(PCI/PCIexpress/EC/non-EC) Digium has several different product numbers for standard PCI(TE405P/410/407/412) and a different number for their PCIexpress cards(TE420) where it seems that they have changed their product naming scheme to be more similar to Sangoma's adding a B for the echo-can version of the card. Here's a run-down of the available quad T1 cards from the 3 big players: Digium: - TE405P - PCI 5v-only, NO hardware EC - TE410P - PCI 3v-only, NO hardware EC - TE406P/TE411P - DISCONTINUED - TE407P - PCI 5v-only, Octasic Hardware Echo-cancellation - TE412P - PCI 3v-only, Octasic Hardware Echo-cancellation - TE420 - PCIexpress, NO hardware EC - TE420B - PCIexpress, Octasic Hardware Echo-cancellation Rhino: - R4T1 - PCI, NO hardware EC - R4T1-e - PCIexpress, NO hardware EC - Add-on Octasic Echo Canceller Sangoma: - a104 - PCI, NO Hardware EC - a104X - PCIexpress, NO Hardware EC - a104d - PCI, Octasic Hardware Echo-cancellation - a104dX - PCIexpress, Octasic Hardware Echo-cancellation From what I've read, the IRQ issues are not present any more on the digium cards. Is the echo cancelling hardware comparable ? I need to install the new card in a dell 2850 or 2950 or possibly even a HP DL360. Anyone have some comments on this ? Do not use Dell. I have had issues with both Sangoma and Digium cards on multiple brand-new Dell servers. This is the only vendor that has consistently given me problems with telco-interface cards. Hope that helps, MATT--- fwiw, my heart says I should buy Digium. My head says I should buy Sangoma. Either my head needs to be convinced that the TE406 is technically as good and as reliable as the 104D, or my heart needs to fall out of love with the romantic notion of supporting Digium because of asterisk. Julian We have used the TE405 and te410 in the past, changed to sangoma A102 (because of problems we were having at the time that we now think may have been telco related). Setting up the digium cards seemed simpler (no patching etc, and no extra wanrouter drivers) but the A102 seem very stable and reliable. Julian Julian ___ --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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email for dotr.com has been scanned by MessageLabs __ ___ --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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email for dotr.com has been scanned by MessageLabs
Re: [asterisk-users] Wifi - Managed access points...
Luis, Like Ron, I have tested deploying several different handsets and have been disappointed. I am currently testing a deployment with a DECT system by Aastra that uses multiple access points the talk SIP to Asterisk and DECT to the handset. Being based on DECT they have good battery life and handover of live calls between points is a key feature. Pricing is along the same as what I would pay for High end access points and good handsets. There are systems like this coming out from Aastra, Snom, Polycom/Kirk and probably some others. If I were going to deploy the setup you are talking about I would check this option out before jumping solely on wifi. If you want contact me off list and I'd be happy to visit in more detail. On 10/10/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Hi, I'm working on a Wifi VoIP project specification. It will have almost 8 APs and 20-30 wifi phones. And after some research, I still having some questions ... 1) Are Managed Access Points (and switch controllers) really important to implement good wifi woip (w/ low latency and acceptable handover time) ? 2) What is the difference between (3com WX1200 + 3com AP 3750) and (DES-1228P + DWL-3140AP) ??? 3) 3com says their AP implements WMM ... and DLink says they priorize VoIP traffic based on VLAN ... are those methods the same ? Thank you, Luis A P Barbosa ___ --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 -- Bruce Reeves Nortex Networks ___ --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] G729a codecs + Asterisk 1.4.11
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDNxdqjpLE0HiOBYRAug5AJ4qjE57UcgHEsmAVQFwPSyMn/dyogCeP3qG UKXWhR9ebm2iw2Ao8VLuSEk= =7O/k -END PGP SIGNATURE- ___ --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] Wifi - Managed access points...
Hope you don't mind I jump in here. I am interested in DECT's handover of live calls. My question is, does the IP address on the phone change when moving from on access point to another? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Wednesday, October 10, 2007 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Wifi - Managed access points... Luis, Like Ron, I have tested deploying several different handsets and have been disappointed. I am currently testing a deployment with a DECT system by Aastra that uses multiple access points the talk SIP to Asterisk and DECT to the handset. Being based on DECT they have good battery life and handover of live calls between points is a key feature. Pricing is along the same as what I would pay for High end access points and good handsets. There are systems like this coming out from Aastra, Snom, Polycom/Kirk and probably some others. If I were going to deploy the setup you are talking about I would check this option out before jumping solely on wifi. If you want contact me off list and I'd be happy to visit in more detail. On 10/10/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Hi, I'm working on a Wifi VoIP project specification. It will have almost 8 APs and 20-30 wifi phones. And after some research, I still having some questions ... 1) Are Managed Access Points (and switch controllers) really important to implement good wifi woip (w/ low latency and acceptable handover time) ? 2) What is the difference between (3com WX1200 + 3com AP 3750) and (DES-1228P + DWL-3140AP) ??? 3) 3com says their AP implements WMM ... and DLink says they priorize VoIP traffic based on VLAN ... are those methods the same ? Thank you, Luis A P Barbosa ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Asterisk behind Multi-NAT question
Michiel van Baak wrote: On 16:32, Tue 09 Oct 07, Steve Totaro wrote: For a small investment of time and money, you can setup OpenVPN and have your own network with no NAT issues whatsoever. That would be my first choice over IAX. Or wait till the ipv6 branch is ready for production. NO MORE NAT ! YAY! I have been holding my breath since IPv6 was the Next Big Thing several years ago. I don't think anyone wants to wait another decade. Thanks, Steve ___ --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] DS3 Interface
Andrew Kohlsmith wrote: On Tuesday 09 October 2007 14:32:38 Matt wrote: http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm And your point, precisely, is what? Someone who has a criminal record can't be a technical authority? Someone can't have a criminal record without being a scumbag? Or perhaps that you prefer to write off those who can best your technical prowess by any means possible? My money's on the latter. -A. FREE KEVIN!!! Oh wait, he is free... Thanks, Steve ___ --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] G729a codecs + Asterisk 1.4.11
Hi: You can check the next command: show g729 and you should see some like this 0/0 encoders/decoders of 2 licensed channels are currently in use or the command show translation or check the asterisk log may be the module is not for you processor version. Best Regards At 09:06 a.m. 10/10/2007, you wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDNxdqjpLE0HiOBYRAug5AJ4qjE57UcgHEsmAVQFwPSyMn/dyogCeP3qG UKXWhR9ebm2iw2Ao8VLuSEk= =7O/k -END PGP SIGNATURE- ___ --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 -- RafaelCanchola Product Development Engineer, FonetGlobal Inc. [EMAIL PROTECTED] http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph ___ --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] IAX2 Trunking behind firewall with no inbound rules
I have an Asterisk box behind a firewall at home with an IAX2 trunk to a provider. When I loose the Internet connection I have to perform an iax2 reload to bring the trunk back up. This is because of the firewall configuration. I do not have a port translation through the firewall so that the IAX2 trunk provider can make a connection into me and I would really like to keep it that way. Is there a way to setup the iax2 configuration so that if it looses the connection it can automatically try to re-establish the trunk? -- Ray ___ --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] Registering Multiple SIP Accounts on One Server to Another Server
Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: Steve Totaro wrote: I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it worked fine except for audio issues that I believe are directly related to IAX2 in version 1.2.x. I have four PRIs and want a separate context for each going into the PBX. This worked very well with IAX. I want to use SIP to see if the audio issues are eliminated but Asterisk does not seem to like multiple SIP account from one box to another (four to be exact) I found this http://www.voip-forum.com/news.php?p=187 which makes me think this is a known problem. Unfortunately, the link goes to an error page. I have tried ever combination of credentials and setting in SIP conf but the calls still fail. I tried friend, user, insecure=very, username, from user, and anything else I could think of. Is there something I am missing or a workaround for this issue? PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP-- PBX (calls fail) PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf-- PBX (calls work) Thanks, Steve Totaro I think I may have figured out my own issue. Since I am creating multiple SIP peers on two boxes that point to each other, I need to define separate ports for each one. Anyone know if that is the case? Makes sense to me but I cannot try it on the live server and my dev boxes are all doing other things. no. It might be the case if you had multiple SIP clients behind the same NAT router connection to a non-local Asterisk box. The userid and password that is sent with the call should make it hit the correct sip.conf entry. Perhaps you are doing something silly in your sip.conf configs. Perhaps I am, let's hope so. This was my latest attempt to get it to work. The other server looks identical except the host IP. [general] ;bindport=5060 bindaddr=0.0.0.0 [default] [span1] type=friend host=192.168.6.2 username=span1 secret= context=to-span1 auth=rsa inkeys=span1-2-fast1 outkey=fast1-2-span1 qualify=yes disallow=all allow=ulaw allow=slin allow=alaw insecure=very I don't use RSA auth so I can't comment on that. My understanding of insecure=very is vague, but you do NOT need it for Asterisk-Asterisk SIP connections and I suspect that is what is causing your problem. I recommend against using qualify. ___ --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] G729a codecs + Asterisk 1.4.11
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ok, I've downloaded the i386 module and it works, I have the module loaded... Thanks for the command!! Rafael Canchola a écrit : Hi: You can check the next command: show g729 and you should see some like this 0/0 encoders/decoders of 2 licensed channels are currently in use or the command show translation or check the asterisk log may be the module is not for you processor version. Best Regards At 09:06 a.m. 10/10/2007, you wrote: Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64/codec_g729a_v32_nocona.tar.gz codec I have registered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.. Any idea how to solve the problem. Thanks Best Regards, Marc LEURENT ___ - --Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users */ Rafael/*/Canchola //*Product Development Engineer*/*, Fonet*Global Inc. [EMAIL PROTECTED] http://www.fonetglobal.com http://www.fonetglobal.com/*Ph. *+ 52 800 022 10 21 ext. 214 + 52 442 167 08 00 *VoIP* 523663899 *d00d! *cyberalph ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDO2XqjpLE0HiOBYRAtcTAJ9YJ8qC83ZxC0+kvf3hfAWvb0/FmgCfb2te F8vtQ07kypElJEsokR1XrD8= =lUtS -END PGP SIGNATURE- ___ --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] DS3 Interface
On 10/9/07, Brian West wrote: [...] All I did was click edit in frontpage and alert them of anonymous publishing priv. were on their servers and they called the FBI [...] I believe you. The astonishing security holes that were engineered by MS so their web editing-publishing-browsing suites could work together were unbelievable. Sometimes the Network Neighborhood included the entire internet :-) For the longest time they lobbied congress to pass laws that would have made it a crime for anyone discovering a security hole, to reveal the hole to the public before MS had a chance to fix it. -baji. -- ___ --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] DS3 Interface
did anyone think about how many concurrent call runs on DS3 and how may call single asterisk instance can handle ?! That board does not have any DSP, Who will do trans-coding ? echo cancellation ? Well, keep us update 2007/10/9, Tim King [EMAIL PROTECTED]: If it hasn't already been done I am looking to put together a team to write drivers for this DS3 card to interface asterisk. http://www.imagestream.com/PCI_921-CDS.html The card itself seems reasonable and I believe we can make it work. As soon as I have positive feedback to begin the project I will put a server on the net with a card in it. Let's make this happen. *Tim King* *CEO* [image: CNS_LOGO_Beveled] http://www.compnetwork.net/ 7589 Cottonwood Drive Suite C Jenison, MI 49428 Phone 616.301.3290Fax: 616.667.1104 Website: http://www.compnetwork.net ___ --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 -- Giovanni Miano image001.png___ --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] Call recording on demand...
Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --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] Wifi - Managed access points...
Wai, The IP address is really on the access points, since they are the SIP part of the solution. Let me see how well I can explain this, The access points register to a manager application, running on one AP, and the phones have a hard coded DECT id and register to the same manager app. The manager actually performs the connection to the Asterisk system and all the access points have IP's and each phone an account on Asterisk. In a handover the manager app routes the SIP traffic to the AP that the handset is on and as the caller moves the phone detects other AP's and picks a new AP. That AP coordinates with the manager app to re-route the sip/rtp. In testing so far, you cannot tell the hand off occurred, even while watching the signal meters on the phone, there is no noticeable audio loss. There has to be a fair overlap in coverage, they say around -60db to -70db in signal you should have another AP and the phone can see up to 4 APs at a time. Each AP can handle 8 voice channels so you have to keep that in mind also. So did that make sense. == Little Commercial blip == Aastra requires people be certified resellers on this solution to purchase\sale it. In that process they give a fantastic amount of attention to planning a wireless deployment. Nortex is a certified reseller of the Aastra SPI-DECT solution. ==End of blip == On 10/10/07, Wai Wu [EMAIL PROTECTED] wrote: Hope you don't mind I jump in here. I am interested in DECT's handover of live calls. My question is, does the IP address on the phone change when moving from on access point to another? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Wednesday, October 10, 2007 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Wifi - Managed access points... Luis, Like Ron, I have tested deploying several different handsets and have been disappointed. I am currently testing a deployment with a DECT system by Aastra that uses multiple access points the talk SIP to Asterisk and DECT to the handset. Being based on DECT they have good battery life and handover of live calls between points is a key feature. Pricing is along the same as what I would pay for High end access points and good handsets. There are systems like this coming out from Aastra, Snom, Polycom/Kirk and probably some others. If I were going to deploy the setup you are talking about I would check this option out before jumping solely on wifi. If you want contact me off list and I'd be happy to visit in more detail. On 10/10/07, Luis Antonio Prata Barbosa [EMAIL PROTECTED] wrote: Hi, I'm working on a Wifi VoIP project specification. It will have almost 8 APs and 20-30 wifi phones. And after some research, I still having some questions ... 1) Are Managed Access Points (and switch controllers) really important to implement good wifi woip (w/ low latency and acceptable handover time) ? 2) What is the difference between (3com WX1200 + 3com AP 3750) and (DES-1228P + DWL-3140AP) ??? 3) 3com says their AP implements WMM ... and DLink says they priorize VoIP traffic based on VLAN ... are those methods the same ? Thank you, Luis A P Barbosa ___ --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 -- Bruce Reeves Nortex Networks ___ --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 -- Bruce Reeves Nortex Networks ___ --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] get egress SIP call Id
Yes, Ben you are right. Asterisk is a B2BUA. When the call passes
through the ingress and egress sip call ids are different. By using
$SIPCALLID I can easily get the sip call id that User A sends. The
question is how to accessing SIP callid of the INVITE sent to User B?
By senting Manager interface channel query commands I can get the egress
sip call id but it is not that easy. Just want to know if there is any a
simple way to do that. Thanks a lot. Ray
Hello Steve,
I think Ray was talking more like the following setup (do correct me
if
I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case, the INVITE SIP callId received by Asterisk from User A
is
different to that sent in the INVITE to User B.
I can get User A's callId using ${SIPCALLID}. How about accessing SIP
callid of the INVITE sent to User B??
Typical need for this, is to store both the callIds to store in the
CDRs
for debugging purposes(w.r.t. the service provider, et al).
cheerz
- Ben.
Steve Totaro wrote:
You can capture the sipcallid from the manager output. The cool
part is that the sipcallid is the same on both sides of a call.
So, AsteriskA---SIP (sipcallid) AsteriskB SIP (Same sipcallid
as AsteriskA for that call.
It is really easy to capture it from the manager.
Thanks,
Steve
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Re: [asterisk-users] Call recording on demand...
Look at features.conf /b On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --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] How to order audio codecs...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have license for g729a audio codecs and I would like user to use them and when the limit of 10 is reached, I would like the others to use ulaw... Do youu know how to do it... I have put: allow=g729,ulaw disallow=all But ulaw is always chosen Have a nice day -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDPePqjpLE0HiOBYRAvSWAJ9Z7gJMDuTw9EcL5of35SmF1slwIwCeM8n/ MfjqNU/3gkdLwKqo1tN5yV8= =3oU/ -END PGP SIGNATURE- ___ --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] Click to Talk Web Applications with Asterisk
On 10/9/07, Senad Jordanovic [EMAIL PROTECTED] wrote: zoachien wrote: Google for mexuar. Zoa Or look at one that works with MS Windows, Linux or Apple http://www.bicomsystems.com/products/C/P/319/382/ FYI, Mexuar's solution -- Corraleta SDK -- *works* with win, linux and mac, from direct experience. What's not so clear from the OP is what is meant by click-to-call: a) Automated dialing solutions via PSTN ? b) Call via a web embedded soft-phone ? (this would be Mexuar) -- exvito ___ --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] Click to Talk Web Applications with Asterisk
On Oct 10, 2007, at 11:12 AM, Ex Vito wrote: On 10/9/07, Senad Jordanovic [EMAIL PROTECTED] wrote: zoachien wrote: Google for mexuar. Zoa Or look at one that works with MS Windows, Linux or Apple http://www.bicomsystems.com/products/C/P/319/382/ FYI, Mexuar's solution -- Corraleta SDK -- *works* with win, linux and mac, from direct experience. What's not so clear from the OP is what is meant by click-to-call: a) Automated dialing solutions via PSTN ? b) Call via a web embedded soft-phone ? (this would be Mexuar) -- exvito I think what he wants is something that does third party call control (3pcc). WeSIP is one but you can't use it in a commercial application without paying for a license. FreeSWITCH can be controlled with 3pcc also and its free. That is what most if not all Click-to-Dial applications use. RFC3725 covers this. http://en.wikipedia.org/wiki/3pcc for more information. /b ___ --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] How to order audio codecs...
if you have allow=g729,ulaw and you want to use g729 but the current channel is ulaw it will pick ulaw over g729 because it wants to escape doing any transcoding if possible. The best way to do this is setup different peers with different allow lines to force the outbound leg to the codec you wish. /b On Oct 10, 2007, at 11:02 AM, Marc LEURENT wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have license for g729a audio codecs and I would like user to use them and when the limit of 10 is reached, I would like the others to use ulaw... Do youu know how to do it... I have put: allow=g729,ulaw disallow=all But ulaw is always chosen Have a nice day -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHDPePqjpLE0HiOBYRAvSWAJ9Z7gJMDuTw9EcL5of35SmF1slwIwCeM8n/ MfjqNU/3gkdLwKqo1tN5yV8= =3oU/ -END PGP SIGNATURE- ___ --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] AST-2007-022: Buffer overflows in voicemail when using IMAP storage
Asterisk Project Security Advisory - AST-2007-022 ++ | Product | Asterisk | |+---| | Summary | Buffer overflows in voicemail when using IMAP | || storage | |+---| | Nature of Advisory | Remotely and locally exploitable buffer overflows | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Minor | |+---| | Exploits Known | No| |+---| |Reported On | October 9, 2007 | |+---| |Reported By | Russell Bryant [EMAIL PROTECTED] | || | || Mark Michelson [EMAIL PROTECTED]| |+---| | Posted On | October 9, 2007 | |+---| | Last Updated On | October 10, 2007 | |+---| | Advisory Contact | Mark Michelson [EMAIL PROTECTED]| |+---| | CVE Name | | ++ ++ | Description | The function sprintf was used heavily throughout the | | | IMAP-specific voicemail code. After auditing the code, | | | two vulnerabilities were discovered, both buffer | | | overflows. | | | | | | The following buffer overflow required write access to | | | Asterisk's configuration files in order to be exploited. | | | | | | 1) If a combination of the astspooldir (set in | | | asterisk.conf), the voicemail context, and voicemail | | | mailbox, were very long, then there was a buffer | | | overflow when playing a message or forwarding a message | | | (in the case of forwarding, the context and mailbox in | | | question are the context and mailbox that the message| | | was being forwarded to). | | | | | | The following buffer overflow could be exploited | | | remotely.| | | | | | 2) If any one of, or any combination of the Content-type | | | or Content-description headers for an e-mail that| | | Asterisk recognized as a voicemail message contained | | | more than a 1024 characters, then a buffer would | | | overflow while listening to a voicemail message via a| | | telephone. It is important to note that this did NOT | | | affect users who get their voicemail via an e-mail | | | client. | ++ ++ | Resolution | sprintf calls have been changed to snprintf wherever | || space was not specifically allocated to the buffer prior | || to the sprintf call. This includes places which are not | || currently prone to buffer overflows. |
[asterisk-users] Asterisk 1.4.13 Released
The Asterisk Development Team has released version 1.4.13. This release fixes a couple of security issues in the implementation of IMAP storage for voicemail. One of the issues is remotely exploitable. Any systems that do not use IMAP storage for voicemail are not affected by these issues. For more details on this issue, see the Asterisk security advisory here: * http://downloads.digium.com/pub/asa/AST-2007-022.pdf This release also contains some other bug fixes that have been merged in the past week or so. The other fixes include resolutions for a few different deadlocks, a couple of problems in res_jabber, chan_sip and RTP fixes, and a few more minor issues. See the ChangeLog for a full listing of the changes: * http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.4.13 Thank you very much for your support! ___ --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] Call recording on demand...
Ok. I know you have to use touch monitor but what I am after is the variables that need to be specified and where in the extensions.conf to configure for users? Brian West [EMAIL PROTECTED] 10/10/2007 12:00 PM Look at features.conf /b On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --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] Wifi - Managed access points...
Thanks. It make perfect sense. I was just curious why the manager app is needed. Since the phone can see 4 AP at the same time, when it wants a call to be handed over to a different AP, couldn't it just send a re-invite to Asterisk and call it a day? Wai, The IP address is really on the access points, since they are the SIP part of the solution. Let me see how well I can explain this, The access points register to a manager application, running on one AP, and the phones have a hard coded DECT id and register to the same manager app. The manager actually performs the connection to the Asterisk system and all the access points have IP's and each phone an account on Asterisk. In a handover the manager app routes the SIP traffic to the AP that the handset is on and as the caller moves the phone detects other AP's and picks a new AP. That AP coordinates with the manager app to re-route the sip/rtp. In testing so far, you cannot tell the hand off occurred, even while watching the signal meters on the phone, there is no noticeable audio loss. There has to be a fair overlap in coverage, they say around -60db to -70db in signal you should have another AP and the phone can see up to 4 APs at a time. Each AP can handle 8 voice channels so you have to keep that in mind also. So did that make sense. -- Bruce Reeves Nortex Networks attachment: winmail.dat___ --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] Call recording on demand...
And is there a way the automon can send the result to voicemail? I hadn't found that yet. Moj Reggie Payne wrote: Ok. I know you have to use touch monitor but what I am after is the variables that need to be specified and where in the extensions.conf to configure for users? Brian West [EMAIL PROTECTED] 10/10/2007 12:00 PM Look at features.conf /b On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg ___ --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] Registering Multiple SIP Accounts on One Server to Another Server
Eric ManxPower Wieling wrote: Steve Totaro wrote: Eric ManxPower Wieling wrote: Steve Totaro wrote: Steve Totaro wrote: I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it worked fine except for audio issues that I believe are directly related to IAX2 in version 1.2.x. I have four PRIs and want a separate context for each going into the PBX. This worked very well with IAX. I want to use SIP to see if the audio issues are eliminated but Asterisk does not seem to like multiple SIP account from one box to another (four to be exact) I found this http://www.voip-forum.com/news.php?p=187 which makes me think this is a known problem. Unfortunately, the link goes to an error page. I have tried ever combination of credentials and setting in SIP conf but the calls still fail. I tried friend, user, insecure=very, username, from user, and anything else I could think of. Is there something I am missing or a workaround for this issue? PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP-- PBX (calls fail) PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf-- PBX (calls work) Thanks, Steve Totaro I think I may have figured out my own issue. Since I am creating multiple SIP peers on two boxes that point to each other, I need to define separate ports for each one. Anyone know if that is the case? Makes sense to me but I cannot try it on the live server and my dev boxes are all doing other things. no. It might be the case if you had multiple SIP clients behind the same NAT router connection to a non-local Asterisk box. The userid and password that is sent with the call should make it hit the correct sip.conf entry. Perhaps you are doing something silly in your sip.conf configs. Perhaps I am, let's hope so. This was my latest attempt to get it to work. The other server looks identical except the host IP. [general] ;bindport=5060 bindaddr=0.0.0.0 [default] [span1] type=friend host=192.168.6.2 username=span1 secret= context=to-span1 auth=rsa inkeys=span1-2-fast1 outkey=fast1-2-span1 qualify=yes disallow=all allow=ulaw allow=slin allow=alaw insecure=very I don't use RSA auth so I can't comment on that. My understanding of insecure=very is vague, but you do NOT need it for Asterisk-Asterisk SIP connections and I suspect that is what is causing your problem. I recommend against using qualify. Thanks, I will give those recommendations a try. If not, I am going to re-do their entire setup in a dev environment and then just move it over after testing. Thanks, Steve totaro ___ --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] Opinions on Release Numbering
I have been having discussions with various members of the development community in regards to changes to the way we manage open source Asterisk releases. The changes that we eventually decide on will determine how we manage the 1.6 version of Asterisk. I will be posting much more detailed information about 1.6 in the near future. What I'm looking for right now is some opinions on version numbering. Part of the working plan for Asterisk 1.6 involves making release candidates for every 1.6.X release, so that various community members can help with doing regression testing on the changes before making the release. I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. What is your opinion? I certainly want the release naming to be as obvious as possible. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] Call recording on demand...
On 10/10/2007, Reggie Payne [EMAIL PROTECTED] wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg Are you suggesting that all of a call is recorded and if a certain key sequence is not entered during the call, the recording is completely discarded otherwise the complete call is saved. Or are you suggesting the call is only recorded from the point you enter a specific key sequence? Ray ___ --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] Opinions on Release Numbering
I second calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. ___ --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] Opinions on Release Numbering
Russell Bryant wrote: I have been having discussions with various members of the development community in regards to changes to the way we manage open source Asterisk releases. The changes that we eventually decide on will determine how we manage the 1.6 version of Asterisk. I will be posting much more detailed information about 1.6 in the near future. What I'm looking for right now is some opinions on version numbering. Part of the working plan for Asterisk 1.6 involves making release candidates for every 1.6.X release, so that various community members can help with doing regression testing on the changes before making the release. I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. What is your opinion? I certainly want the release naming to be as obvious as possible. I think that using 1.5.x as the name for a release candidate for 1.6 is pretty close to as unintuitive as it can possibly be. 1.6.Xrc-Y is a strikingly MORE intuitive naming scheme for 1.6 release candidates. N. ___ --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] Opinions on Release Numbering
Russell Bryant wrote: I have been having discussions with various members of the development community in regards to changes to the way we manage open source Asterisk releases. The changes that we eventually decide on will determine how we manage the 1.6 version of Asterisk. I will be posting much more detailed information about 1.6 in the near future. What I'm looking for right now is some opinions on version numbering. Part of the working plan for Asterisk 1.6 involves making release candidates for every 1.6.X release, so that various community members can help with doing regression testing on the changes before making the release. I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. yes for me. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. eek. no. What is your opinion? I certainly want the release naming to be as obvious as possible. Julian ___ --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] Opinions on Release Numbering
Russell Bryant wrote: I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. What is your opinion? I certainly want the release naming to be as obvious as possible. Then I think that would be the rc1,rc2 style then. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Call recording on demand...
The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Razza [EMAIL PROTECTED] 10/10/2007 1:56 PM On 10/10/2007, Reggie Payne [EMAIL PROTECTED] wrote: Hello All! I am new to the list. Does know how to record a call on demand? What I would like to do is setup something that during a call someone can hit a button a the call is recorded the after the call is over the recording is sent to their voicemail. Anyone? Thanks, Reg Are you suggesting that all of a call is recorded and if a certain key sequence is not entered during the call, the recording is completely discarded otherwise the complete call is saved. Or are you suggesting the call is only recorded from the point you enter a specific key sequence? Ray ___ --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] Opinions on Release Numbering
On Wednesday October 10 2007 2:15 pm, Doug Lytle wrote: Russell Bryant wrote: I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. What is your opinion? I certainly want the release naming to be as obvious as possible. I would say the rc-1, rc-2 is about as obvious as it gets and would get my vote. JohnM -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 Phone (603) 764-9163 ___ --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] Opinions on Release Numbering
On Wed, Oct 10, 2007 at 12:54:42PM -0500, Russell Bryant wrote: What is your opinion? I certainly want the release naming to be as obvious as possible. Wikipedia has something to say on this (by which, of course, I mean me :-)... The traditional approach to this is, roughly 1.5.8 1.5.9 1.5.10 1.5.11 == 1.6a1 1.6a2 1.6a3 1.6a4 == 1.6b1 1.6b2 1.6b3 1.6b4 == 1.6rc1 1.6rc2 1.6rc3 == 1.6.0 1.6.1 1.6.2 ... The important points (IME) are these: 1) the first release of a transition level is exactly equivalent to the differently numbered release it replaces. This is most important coming out of Release Candidates: you *must not make any changes* between your last RC and your production release. If you do, it's really another beta. (The common distinction between betas and RC's is that betas are permitted new features, but RC's come after the feature freeze, and aren't.) 2) If you promote a level, and it turns out not to be robust enough to support it, you can either demote it and try again, or just march ahead and fix the bugs, but you can't reuse a version number for different code. 3) Version numbers serve 2 purposes: they're a contract with the user about the expectations they can have reasonably about the release as it sits -- if I see something that's an RC2 coming off 5 betas, then I can make some assumptions about how stable and reliable I think that code's likely to be -- if the release manager hasn't een playing fast and loose with the numbering. (Specifically, if you make any changes between your last beta and your first RC, then it's not really an RC; it's another beta.) And secondly, they're a contract between users and technical support, so that TS knows *exactly* what code base the user has and can debug problems reliably -- which is even more critical in the open source world where your TS team is other users than it is in commercial software. Just my thoughts from observation of 25 years of development... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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] Opinions on Release Numbering
rc1, rc2 is the best choice for me. Best Regards. Emiliano Vazquez. - Original Message - From: Russell Bryant [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 10, 2007 2:54 PM Subject: [asterisk-users] Opinions on Release Numbering I have been having discussions with various members of the development community in regards to changes to the way we manage open source Asterisk releases. The changes that we eventually decide on will determine how we manage the 1.6 version of Asterisk. I will be posting much more detailed information about 1.6 in the near future. What I'm looking for right now is some opinions on version numbering. Part of the working plan for Asterisk 1.6 involves making release candidates for every 1.6.X release, so that various community members can help with doing regression testing on the changes before making the release. I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. What is your opinion? I certainly want the release naming to be as obvious as possible. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] Echo Problems with IAXy
Typically, echo isn't heard on the _far_ end, unless it is created by acoustic effects within the phone hooked up to the IAXy. Can the microphone hear the speaker? You said you've tried numerous analog phones, so that kind of rules that out, but curious... Sean Dennis wrote: From what I have found the IAXy doesn't handle echo very well. About half of the analog phones I try on the adapter create an echo on the far end. The person I am talking to can hear themselves. I am using Asterisk 1.4 and have tried it with 1.2 as well with the same results. Is there is anything I can do in Asterisk to help solve the echo problem? Thanks, Sean Dennis ___ --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] Opinions on Release Numbering
Russell Bryant wrote: I have been having discussions with various members of the development community in regards to changes to the way we manage open source Asterisk releases. The changes that we eventually decide on will determine how we manage the 1.6 version of Asterisk. I will be posting much more detailed information about 1.6 in the near future. What I'm looking for right now is some opinions on version numbering. Part of the working plan for Asterisk 1.6 involves making release candidates for every 1.6.X release, so that various community members can help with doing regression testing on the changes before making the release. I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. What is your opinion? I certainly want the release naming to be as obvious as possible. If I remember what was discussed in a recent VoIP users conference, you guys (being digium) were considering moving to a more rapid release schedule similar to how the linux kernel is currently released. IE 1.6.4 would likely contain additional features over 1.6.3 and 1.6.3.1 would contain bug fixes for 1.6.3. That being the case I think the 1.5.x scheme would get confusing very quick. Example: is 1.5.3.1 the second RC for 1.6.3 or the first RC for 1.6.3.1? I would vote for the 1.6.3.x-rc1,rc2 etc scheme. This does begs the question of the purpose of the odd number releases 1.1.x,1.3.x,1.5.x (which don't exist). Will asterisk continue to increment in even number releases just because or will odd numbers be used at some point? -Dave ___ --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] Randomly half-voice at sip/zap
Péter Tóth wrote: When i try ztmonitor as follows, it gives strange output... [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* R ###* R If ztmonitor keeps scrolling down the screen, you need to make your terminal wider. The '#' marks should jump back and forth left and right like a level monitor, and there will only be one row of them (but with two levels, one for RX and one for TX). The screen won't scroll at all. Try this again :) Moj ___ --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] Call recording on demand...
Reggie Payne wrote: The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Except for the sending to voicemail bit, I have some scripts I put together at http://horanappraisals.com/asterisk/recordings/ that provide a simple web interface to asterisk's recordings directory. Depending on the version of asterisk installed, the parsing of the name of the monitor filename might be a little off, but it shouldn't be hard to straighten out. Moj ___ --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] Call recording on demand...
Reggie Payne wrote: The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Use a script run regularly from cron to detect new recordings in the monitor directory, determine who the recipient should be, and mail them out. ___ --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] How are you using Asterisk at Home ?
I setup Trixbox on an Dell Precision 360. I ported my old POTS line over to a pay-as-you-go through Teliax because we weren't using more than 500 minutes a month on the home line. When a caller rings in, I screen the call with time-of-day routing. In general, if the call comes before 7:30 AM or after 10:30 PM, it isn't going to ring through (we had 'problems' with my father-in-law calling us at 7:00 on Saturday to see what we were doing). Instead, they get a voice menu with me politely telling the caller we're not accepting calls at that time. But, I added a code of '111' to that menu and gave it to the family. If they are calling with an emergency, they enter that code and it rings all the extensions in the house plus both of our cell phones. The first one to pickup grabs the call. If calls aren't restricted by TOD, they have to get past privacy manager and blacklist before they will ring some of the extensions (did this with a ring group). If nobody picks up, they are dropped into a voice menu that allows them to leave either of us messages or transfer to our cell phones. This way we can just give everyone a single number and not worry about letting out our cell phone numbers. Of course, calls to the cell phones are confirmed so when one comes in, we have to hit 1 on the cell if we want to accept the call... otherwise its back into VM for the caller. Of course, voicemails are sent via e-mail to my wife and I and I also setup an Aastra 57i on my desk at work that connects to the company server on line 1 and to the home box on line 2. I even got a second line from Teliax in August and set it up to only ring the phone at work. I used this line while I was setting up the wife's surprise 30th birthday party. It was brilliant because guests could call me and there was no trace of the call on my cell phone where she might see it and it didn't ring the home phones. I'm not doing anything really cool like pausing the TV but the setup has worked very well and has given us control over the phone. Instead of us being slaves to when people call, they get through at our pleasure now. It has been a big improvement. (plus it has impressed some of my friends!) Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of D4rk F1ber Sent: Monday, October 08, 2007 6:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How are you using Asterisk at Home ? I am very new to Asterisk, it was a weekend project of mine that I jumped into this weekend. I have it up and working on a box at home, and I am nearly half way through the book I purchased friday Asterisk: The Future of Telephony 2nd Edition. Anyway, I started this out so I could help a friend who wanted a VoIP PBX solution for his small business. I have been working with Cisco Callmanager for about 6 years now, and prior to that did help manage other PBXs as well as work on various Motorola VoFR projects as well. My friend came to me and well everything I deal with is really for larger businesses, and since I had heard about Asterisk in the past I thought it would be a good reason to finally jump into it. And what a jump it has been. Only scratching the surface with this thing and well I am very impressed with what I have seen so far. The main point for me writting others is to find out how others are using Asterisk for the home? Bit of over kill for most I am sure, and to be honest we (Wife, kid and I) don't even have a home phone anymore. After playing with this though, shesh I could have fun with it at home. :-) Thinking about getting a SIP line or trunk or something to tie into this for home usage. One of the next projects for me personally is to get a SIP client for my Cingular/ATT 8525, it has wifi and hsdpa running Windows Mobile 6 and I am certain I have run across SIP clients before for these things. Be fun to play with and get working. So yes I am asking because I am unimaginative and need ideas on selling this to the wife. :-) That and I am just curious about what others feel are useful uses for it within the home, and what others get excited about regarding it all. ___ --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] Opinions on Release Numbering
On Wed, Oct 10, 2007 at 02:10:54PM -0400, SIP wrote: [snip] I think that using 1.5.x as the name for a release candidate for 1.6 is pretty close to as unintuitive as it can possibly be. 1.6.Xrc-Y is a strikingly MORE intuitive naming scheme for 1.6 release candidates. mutt uses the x.y convention where y is odd for a development branch and y is odd for a release branch. So 1.5 would be the development of 1.4 etc. When it's stable a 1.6 would be released which would only have bug/security releases, any new features etc would go into 1.7. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Call recording on demand...
Awesome. Thanks all. I am still gonna work on some other possible logic. It would really be cool to have all of that functionality in Asterisk. Reg Mojo with Horan Company, LLC [EMAIL PROTECTED] 10/10/2007 3:24 PM Reggie Payne wrote: The call is recorded after a key sequence has been pressed. Example: SIP/101 makes an outbound call to 5551212 5551212 starts to get rowdy SIP/101 enters *99 to start recording the call After the call is ended the recording is sent to the voicemail of 101 Except for the sending to voicemail bit, I have some scripts I put together at http://horanappraisals.com/asterisk/recordings/ that provide a simple web interface to asterisk's recordings directory. Depending on the version of asterisk installed, the parsing of the name of the monitor filename might be a little off, but it shouldn't be hard to straighten out. Moj ___ --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] How are you using Asterisk at Home ?
GNUbie wrote: By the way, my Asterisk PBX server is also my wireless access point, web server, file server, music server, VPN server, database server, firewall and router. Repeat after me - NEVER NEVER NEVER run other servers on your router/firewall machine!!! That machine needs to be a maximum security low vulnerability box and running all sorts of stuff on it conflicts with that. Your web server is probably your weakest link in security, so I wouldn't put your file server, music server, or database server on that same box because if someone hacks through some webapp you've installed (it's happened to me with both the TWiki and awstats packages) then if they've got root on your web server box you don't want them messing with the other stuff. I know it sounds like overkill, but I see three boxes here: 1 - firewall/router 2 - web server and other public facing services (sendmail for example) 3 - internal facing services - database, asterisk, file/music server Some day when box #2 gets rooted (and it will eventually) you'll thank me... Steve ___ --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] RTP Packets not received from Asterisk
Hi I am new to Asterisk, I am writing a softphone but facing few problems: 1. Call is successfully established between two clients but I am unable to receive RTP packets. All PCs are in same network domain. One of the client is X-lite and other client is my softphone. Vinay ___ --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] Bug #0010567, any news?
Hi guys, I'm not sure here is the best place to ask, but, anyone has some news regarding to this bug? I'm having problems with this in one customer. Thanks Carlos Barros ___ --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] Opinions on Release Numbering
My opinion: 1.4 is a branch. current trunk should be called 1.5 1.5 should be 1.5.1.1, 1.5.1.2 ,1.5.1.3,1.5.2 In the above, X.X.Y denotes the stable version. Any changes to that code, would use the next point value. 1.5.1.Z You do not change to 1.5.2.0 until it has been tested, thus 1.5.2 would be the stable release of the last 1.5.1.Z. You could think of it as beta1, Beta2, RC1, RC2, etc. just without all those nasty letter in the version number. You could also drop the 1s and move everything over one spot in my opinion. At a year between releases (not a slam by the way) I think you could use full integer increments on the versions. -- -- Steven http://www.glimasoutheast.org Russell Bryant [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have been having discussions with various members of the development community in regards to changes to the way we manage open source Asterisk releases. The changes that we eventually decide on will determine how we manage the 1.6 version of Asterisk. I will be posting much more detailed information about 1.6 in the near future. What I'm looking for right now is some opinions on version numbering. Part of the working plan for Asterisk 1.6 involves making release candidates for every 1.6.X release, so that various community members can help with doing regression testing on the changes before making the release. I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. What is your opinion? I certainly want the release naming to be as obvious as possible. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] Meetme conference room duplex issue
Hello. We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2). We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex. If a person starts talking they will cut off others on the call. Is this normal behavior? Are there any options I can change to change this? Thanks! James___ --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] Meetme conference room duplex issue
I have not noticed this here at all -- although too much of talking over each other makes a mess, but in both 1.2 and 1.4 I have not noticed any such behavior. What are you using for a carrier? on Wednesday 10/10/2007 jamespev([EMAIL PROTECTED]) wrote Hello. We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2). We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex. If a person starts talking they will cut off others on the call. Is this normal behavior? Are there any options I can change to change this? Thanks! Jamesbr / nbsp;nbsp; Hello.nbsp; We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).nbsp; We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.nbsp; If a person starts talking they will cut off others on the call.nbsp; Is this normal behavior?nbsp; Are there any options I can change to change this?br / br / nbsp;nbsp; Thanks!br / br / Jamesbr / br / ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] How are you using Asterisk at Home ?
Nonsense! I'm a Security Expert (TM) and I say run EVERYthing on your firewall And...uh... what was your IP again? ;) N. Steve Prior wrote: GNUbie wrote: By the way, my Asterisk PBX server is also my wireless access point, web server, file server, music server, VPN server, database server, firewall and router. Repeat after me - NEVER NEVER NEVER run other servers on your router/firewall machine!!! That machine needs to be a maximum security low vulnerability box and running all sorts of stuff on it conflicts with that. Your web server is probably your weakest link in security, so I wouldn't put your file server, music server, or database server on that same box because if someone hacks through some webapp you've installed (it's happened to me with both the TWiki and awstats packages) then if they've got root on your web server box you don't want them messing with the other stuff. I know it sounds like overkill, but I see three boxes here: 1 - firewall/router 2 - web server and other public facing services (sendmail for example) 3 - internal facing services - database, asterisk, file/music server Some day when box #2 gets rooted (and it will eventually) you'll thank me... Steve ___ --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] How are you using Asterisk at Home ?
If all the services are for internal use and authorized external use then there would be no problem with doing this. Deny all ports on the external facing interface except 1194 or whatever you want to run OpenVPN on and you can connect remotely over the VPN and be totally safe from the outside world. You could also open up SSH and use tunneling for your needs. Thanks, Steve SIP wrote: Nonsense! I'm a Security Expert (TM) and I say run EVERYthing on your firewall And...uh... what was your IP again? ;) N. Steve Prior wrote: GNUbie wrote: By the way, my Asterisk PBX server is also my wireless access point, web server, file server, music server, VPN server, database server, firewall and router. Repeat after me - NEVER NEVER NEVER run other servers on your router/firewall machine!!! That machine needs to be a maximum security low vulnerability box and running all sorts of stuff on it conflicts with that. Your web server is probably your weakest link in security, so I wouldn't put your file server, music server, or database server on that same box because if someone hacks through some webapp you've installed (it's happened to me with both the TWiki and awstats packages) then if they've got root on your web server box you don't want them messing with the other stuff. I know it sounds like overkill, but I see three boxes here: 1 - firewall/router 2 - web server and other public facing services (sendmail for example) 3 - internal facing services - database, asterisk, file/music server Some day when box #2 gets rooted (and it will eventually) you'll thank me... Steve ___ --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] Meetme conference room duplex issue
Are you using zap channels with 'aggressive' echo suppression enabled? That will make calls pretty half-duplex. Moj jamespev wrote: Hello. We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2). We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex. If a person starts talking they will cut off others on the call. Is this normal behavior? Are there any options I can change to change this? Thanks! James ___ --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] Distributed FAX - How to best complement asterisk ?
Hi list, I'm evaluating a private telephony scenario of about 20 locations - 300 phones, 50 FAX machines. Initial overview points to the installation of asterisk at three locations connected to the PSTN via ISDN PRI. All other locations, small by themselves, would get SIP phones managed by asterisk, since there is good IP connectivity between all sites. Now on to the subject... Handling FAXes: 1. On the locations where asterisk is installed, the solution is trivial; either by connecting FAXes to FXS ports on channelbanks or by managing faxes with iaxmodem + Hylafax. Probably a combination of both... 2. On the remaining locations we have a problem which I have been studying and trying to address... Faxing over IP. Side note: - I've read every recent mail on this mailing list regarding the subject - I've browsed the wiki to its fullest extent - I've googled a lot - I've read Steve Underwood's excelent summary on the subject (check it out at http://www.soft-switch.org/foip.html) Facts: a) FAX over VoIP will not work, so installing ATAs on the remote locations and bridging them with the PSTN FAXes is out of the plan. b) T.38 is the answer to FoIP c) asterisk 1.2 does not support T.38 d) asterisk 1.4 only does T.38 passthrough, not good enough e) CallWeaver seems to support T.38 gatewaying, although I'd rather move on with asterisk so as to leverage current experience and knowledge and to keep installed base with the same software. Possible solutions point to complementing asterisk installations with T.38 capable equipment. (of course, one other solution would be to subscribe to analog lines at each location! however, this would prevent us from performing FAX CDR accounting -- not a requirement, but a really nice-to-have). Having said all of this (and please correct me if I'm wrong) I'm looking for suggestions on how to best complement asterisk in such a scenario. The architecture I'm currently considering is: [PSTN] ---PRI--- [asterisk] ---PRI--- [PRI-to-T38 GW] ... ... --SIP/T.38--- [T.38 ATA] ---FXS--- FAX machine On incoming faxes, asterisk would simply Dial() the PRI leading to the PRI-to-T.38 GW which would be configured, according to the dialed number, to connect via SIP/T.38 to the respective T.38 ATA. Outbound would have the PRI-to-T.38 GW work the other way around, calling asterisk with the PSTN FAX destination number... Again, asterisk would only have to Dial() out to the PSTN. My questions: 1. What do you think of it, Is it feasible ? Does it make any sense ? How would you do it differently and why ? 2. I believe a Cisco AS53xx + Cisco ATAs would do the job. What about a Patton SmartNode 4960 + Patton ATAs ? (I have very little knowledge about Cisco equipment, but I'm almost 100% sure the Ciscos would do it... on the other hand, I've read most of the Patton docs and, again, I'm also almost 100% sure these would do it -- however, hands on experience and knowledge counts a lot!) 3. Roughly, how much would one expect to pay for one such PRI-to-T.38 gateway ? 5k, 10k, 20k ? Probably the Cisco version will be more expensive, no ? 4. Of course, I could use CallWeaver as a PRI-to-T.38 gateway... But then again, how solid would it be ? With which ATAs ? The CallWeaver website shows a very small amount of ATAs confirmed to be 100% working in T.38. 5. Would I need to have a SIP proxy between the PRI-to-T.38 gw and the T.38 ATAs or would they be able to talk to each other directly ? (I'd say this would depend on the specific equipment, but...) If that would be a requirement, which way would you go, asterisk 1.4 ? Would SER forward T.38 traffic ? Thanks for inputs and experiences in complementing asterisk with T.38 equipment. -- exvito ___ --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] shared system - how to monitor channels
I was wondering how everyone here is giving users (say via the BLF on a Polycom, or the sidecart/buddies) the ability to see how many channels they have in their group and how many are in use. Since so many users are used to seeing Line 1, 2, 3 etc on a key system I have been trying to think about how to show channels as a buddy (ie hint). Any suggestions? Bill ___ --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] Paging possible on an ATA?
We've got our Polycom phones auto-answering for paging. Is it possible to configure a PAP2 to auto-answer for either paging or intercom? If so, how? ___ --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] Opinion on hardware (computer) for an Asterisk Server!
Hi list, I'm about to install Asterisk on an Old HP NetServer LC2000 Server (year 2001), it has 2 Pentium III 1GHz CPUs (Coppermine FSB 133MHz 256K L2 Cache), 768MB PC-133 ECC RAM, 3 UltraSCSI LVD2 18.2GB 10K RPM HDD in RAID5, 100Mb NIC for server. This Server will support 35 SIP phones (users) and 10 FXO ports (for telco lines) and 2 FXS ports (internal analog phones) with a Sangoma Remora A400 PCI card. What do you think? is this hardware enough for this setup??? Thanks in advance for your opinions, Raul ___ --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] Paging possible on an ATA?
Is it possible to configure a PAP2 to auto-answer for either paging or intercom? No. You cannot force the connected device (phone) to auto-answer. Imagine you have a plain old phone attached to it, who's going to lift the receiver? ___ --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
