Re: [asterisk-users] Asterisk-Asterisk E1 connection
You need a E1/T1 crossover cable, which isn't straight through or like a network crossover cable. Search online for T1 crossover and you'll find the pinout. Remember one node needs to be the clock source (and only one node). Technically UTP isn't the right cable for E1/T1s, but if your distance between boxes is a couple hundred feet or less, it will work fine. You might also look at TDMoE. Ethernet interfaces are a lot cheaper than E1s. On Apr 11, 2011, at 7:43 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contact Directory on Polycom phones
I use the mini-web browser built into the phone and have a custom button (directory) that accesses the directory, which is hosted on a web server. It isn't perfect, but it's better than the XML files IMHO. That said, there's an enterprise license for these phones which enables directory integration. On Thu, Mar 3, 2011 at 9:20 AM, deeps backup backup.de...@gmail.com wrote: Hi, Polycom phones configured on asterisk pbx and are using contact directory on phones. To modify entries xml file for each phone needs to be modified and have to reboot all phones to accept updated file. Is there any way via asterisk, that we can use central database and on modification automatically update xml files on boot server and reboot phones. Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
My take on this is to not skimp on the phones. This is how people relate to the phone system you install. Good phones will, to them, imply a good system. And vise-versa. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended Windows client to display CID?
I have asterisk call out to a shell script which sends a jabber message to the user (along with links to any open tickets in our ticketing system associated with that CID). All free, but requires work to build. On Jan 26, 2011, at 6:52 AM, Gilles codecompl...@free.fr wrote: Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple clients can connect simultaneously and each be notified of incoming calls. There may be yet other ways to do what I want. Are there open-source solutions you could recommend? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended Windows client to display CID?
On Wed, Jan 26, 2011 at 7:55 AM, Tom Rymes try...@rymes.com wrote:
Ooh. I like this. Can you post a sample, or maybe a synopsis of what pieces
you are using to tie this all together?
I have a two processes - one to notify on an internal incoming call,
one to notify on tickets (both on incoming and outgoing calls).
The notify on incoming call just does the basic CID information. I
have a dialplan line like:
exten = _XXX,1,System(/usr/local/bin/notify_incoming_cid.pl ${EXTEN}
${CALLERID(NUMBER)} ${CALLERID(NAME)} )
This is a Perl script that reads a text file listing extension # and
Jabber ID associations, and, if it finds an association, calls a
second Perl script to send a Jabber notification, using Net::Jabber.
In addition to this, any time a call is placed, a line like the
following executes:
exten = _X.,n,System(/usr/local/bin/notify_it_jira_users.pl
${CALLERID(NUMBER)} ${ext} )
This script uses a similar method to above, but only generates a
notification if a Jira (our ticketting system) user ID associated (via
another text file) with a phone number is a reporter on any open
Jira issues (it does this via a web query to our ticketting system).
If this user is a reporter, the other leg of the call (whether
incoming or outgoing) gets a IM with a link to the specific issues
along with the summary of each issue.
For instance, if someone calls the IT department, they'll get
something like this:
---
303-555-0010 John Smith
---
CALL FROM jsmith WITH TICKETS:
http://jira/IT-1010 - Cannot log into VPN
http://jira/IT-1020 - Computer making strange sounds
---
If they don't have any open tickets, they won't get the second message
listing tickets.
We can generate the text files this solution uses automatically by
looking at our phone list database and our customer database in a Cron
tab (it would be possible to query directly the database, but this was
simpler to implement in an afternoon).
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Re: [asterisk-users] Top Posting
On Tue, Jan 18, 2011 at 8:18 AM, Andrew Thomas a...@datavox.co.uk wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? That's not the alternative (having ten messages above the reply). See this message for an example. I suspect you won't have to scroll at all or read any of the 10+ previous messages. Top posting is here - to stay! It may be. But it would be nice if people cut out the $#@! that is irrelevant to their reply regardless, and were open to hearing what others had to say, rather than saying, I do it this way, it's the best. I also agree this is a pointless discussion because, clearly, nobody is willing to budge, and it has nothing to do with Asterisk. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax_digium.so crashing
On Tue, Jan 18, 2011 at 1:21 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: I got it fixed with an all nighter, but I took a beating for the problems for not fully testing and monitoring. After that, nobody had faith in the fax solution. So is FFA working for you now? What did you have to do to fix it (I like to avoid problems and learning from others is one way to avoid them)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Are there reasons to prefer the use of PRI over SIP or SIP over PRI? Assuming you are talking to connect a PBX to the PSTN... PRI advantages: 1. Relatively little equipment between the PTSN and the PBX. Less to break or go wrong. 2. Simple to set up. No need for QoS, routing, authentication, etc. Of course if you only know IP, SIP is easier, but if you learn both, ISDN is easier. 3. If compared to SIP over internet, PRI has guaranteed quality. Granted, SIP *can* have just as good (and better) quality, just not guaranteed if done over the internet (it can be guaranteed over a private circuit). 4. Less latency/delay so there is less talk-over. 5. FAX, high speed modem, TTY, etc, pass-through actually works. (it *can* work over SIP, but Asterisk just isn't quite there yet) I run the PBX for my organization which has about 160 extensions. I wouldn't even think of doing anything but PRI for the main lines because (A) for our size organization where we are located, we're talking a couple hundred dollars a month difference between PRI and SIP in cost so it's nearly break-even in cost which means cost difference isn't a huge motivator, (B) it supports FAX, modems, and TTYs - perfectly, (C) Quality is 100% consistent. In addition, the reliability is good enough that I'm willing to use it for 911. Of course if this installation wasn't in downtown Denver, where ISDN PRI is very cheap (a full CLEC 23-channel ISDN PRI costs roughly what 6 or 7 ILEC POTS lines cost), then SIP would be interested. SIP advantages: 1. Cheap (at least SIP-over-internet) 2. Easy and quick to scale if you have bandwidth. 3. Great for disaster recovery if using SIP over internet 4. Very cheap to get local numbers from all around the world. 5. If using SIP over internet, easy to compare providers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf CALLERID(num)
Get rid of the spaces before and after the equal sign.
On Wed, Dec 29, 2010 at 4:15 PM, Joseph syscon...@gmail.com wrote:
I'm testing GotoIf($[${CALLERID(num) but I'm missing something as it is not
working:
[office-open]
exten = s,1,Wait(1)
exten = s,2,Answer()
; for Caller ID is 471-5665, always signal congestion:
exten = s,3,GotoIf($[${CALLERID(num)} = 4715665]?4:6)
exten = s,4,Playtones(congestion)
exten = s,5,Congestion(5)
exten = s,6,SetMusicOnHold(default)
...
but it always goes to s,6
What am I missing?
--
Joseph
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Re: [asterisk-users] * 1.8: cannot load g729 free codec (on 1.4 it worked!)
I'm going to guess you aren't going to get a lot of help on a list hosted by Digium on how to use a potentially illegal codec... That said, ast14 in the filename might signify what the problem is. The APIs likely changed for modules between 1.4 and 1.8. On Wed, Dec 22, 2010 at 7:58 AM, Giorgio Incantalupo gincantal...@fgasoftware.com wrote: pbx18*CLI module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]: loader.c:852 load_resource: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' could not be loaded. It worked on Asterisk 1.4, but not anymore on my Asterisk 1.8...why??? :( Thank you Giorgio Incantalupo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem
On Tue, Nov 30, 2010 at 2:28 AM, bilal ghayyad bilmar...@yahoo.com wrote: If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Not necessarily. See below. Basically the problem is that you have a congested link, and TCP is not the fix for congestion. Are you sure you are getting packet loss, and not just delayed packets, that might be arriving AFTER the jitter-buffer's max delay? Either would create the same symptom. But the solution to them is slightly different. Same thing if we used the VPN, and in case of other users are using the Internet to do browsing and downloading then the voice quality will be better than without VPN as the VPN is using TCP? TCP VPNs are bad for several reasons - namely that TCP inside TCP will generate excessive and unnecessary retransmissions. That's why most VPNs use UDP or IPSEC. TCP in TCP will increase delay and/or congestion on your links. The internet bandwidth is not that small .. but the users are doing a big amount of work and we would like to overcome the packets losses in case of using the UDP as the packets are not resend. Any advise for this? Yes. If you are using DSL/cable/other-commodity-circuit, I'd suggest a second DSL circuit to be used only for VoIP. Nobody likes to pay for that, I know, but that's really the solution. If you are using an (expensive) enterprise-class circuit (metro ethernet, DS3, OC3, etc) for internet, work with your provider. At the very least, have the provider does some form of fair queuing and you do the same, you'll probably eliminate 95% of your problem. If they are willing to do QoS to your specs, even better (but I wouldn't count on this). But clearly the way the circuit is configured today, you are having packet loss (the cutting out of voice) or excessive queing of packets. This is because queues in routers are getting too full, and something has to be dropped or something is arriving too late for the jitter buffers on the VoIP equipment to compensate. In otherwords, you are bandwidth constrained. So you need to either increase your bandwidth (expensive!) or implement QoS of some type. There are some ways to implement QoS on your end if your ISP won't cooperate, but it's not a 100% perfect solution. What could be a solution that I can apply it to resolve the voice cutting if the Asterisk was using the internet that is shared with the users in the office that are doing download and browsing? QoS. One more thing, what about using the Buffering or any other technique that can help to overcome packet losses due to the internet download and browsing? Certainly. If your problem is lost packets, you need QoS or bandwidth, but that aside, increased buffers in routers might help or hurt, depending on how things are behaving. You can try both (your ISP will need to do the same, if you are getting cut-outs on inbound packets; if you can get your ISP to adjust this, you can probably get him to just implement QoS and be done with this; If he can't implement QoS, at least get him to do some sort of fair queuing!). If your problem is excessively delayed (due to queuing) packets, you also need QoS or bandwidth. But you can increase the jitter buffer on both ends of the VoIP call. If you use a VoIP provider, they will need to increase the buffer size on their end. Of course this will increase the amount of talk-over and result in less user satisfaction. Delay is a bad thing on phone calls. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording format
WAV or wav? One of these has GSM-encoding inside a WAV formatted envelope. That said, I wouldn't expect that to have any noticeable CPU utilization above that of GSM. If you are using the non-GSM version of WAV, then I am as baffled as you - hopefully someone who knows more about this can help. On Mon, Nov 22, 2010 at 7:58 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi Joel, We have a meetme on which we are landing two G.711 alaw calls, one coming from TDM another from SIP. Once we those parties are in the conference we are adding one more leg using Local channel and starting to record it. Surely it would be logical if it would be less overhead recording alaw wav since we are using alaw on both parties, but its not. Thanks, Vilius. On 22 November 2010 14:19, Joel Maslak jmas...@antelope.net wrote: What format are the actual calls in? Are they in G.711u/a format or are they in something else (perhaps gsm?) format? I'm asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius vilius.adamkavic...@invade.net wrote: Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this? Is Asterisk updating wav headers every time it writes? What would be recommended hardware setup for over 60 simultaneous call records? Regards, Vilius. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Vilius Adamkavicius InVADE Technical Support 3 Berkeley Crescent, Bristol United Kingdom BS8 1HA Company Registration Number: 3660482 Registered in England and Wales this email, and any attachment, is intended only for the attention of the addressee. Its unauthorised use, disclosure, storage or copying is not permitted. If you are not the intended recipient, please destroy all copies and inform the sender by return email. If you have received this email in error, please return it to the sender and highlight the error. We accept no legal liability for the content of the message. Any opinions or views presented are solely the responsibility of the author and do not necessarily represent those of InVADE. We cannot guarantee that this message has not been modified in transit, and this message should not be viewed as contractually binding. Although we have taken reasonable steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. international phone number +44(0) 117 33 555 00 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 re-invites issue
NAT? Firewall? On Thu, Nov 11, 2010 at 3:21 PM, Marek Soha ma...@snet.sk wrote: Hi all. I have an issue with T.38 and re-invites. Topology: provider - A (asterisk 1.6) - B (asterisk 1.6) - extension - - (software fax, gateway whatever). When between A and B trunk is canreinvite=no everything is working smooth. When I switch canreinvite to yes, it stop working. Do you have any idea where the issue can be? Any help will be much appreciated. Marek Soha -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
On Mon, Nov 8, 2010 at 7:05 AM, Cary Fitch ca...@usawide.net wrote: Does anyone know if ATT EELs delivered to a CLEC would be PRI, or Robbed bit? It won't be ISDN. It will be some form of RBS. You probably have several choices as to which type of RBS (probably several ESF options, you'll probably pick one of them; you may be able to use SF as well). You should probably work with your LEC to figure out exactly what they will hand off to you. You might make a costly mistake if you don't. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
I believe this looks like a standard channel bank. Asterisk generates all audio. Ring and hook status are sent out of band. Dial tones are in-band. Ringback, busy, congestion are in-band audio. I would think a standard T1 card would be fine. That said, I would verify this with the LEC. On Nov 7, 2010, at 1:22 PM, Cary Fitch ca...@usawide.net wrote: Alternate question: Asterisk/PSTN oriented. If an Asterisk system were interfaced via a T1 to a local telco loop to a customer premises: (This is not a T1 to the customer premises, but a T1 to the telco who then demuxes it to copper to the customer premises. IE. In Telecom terms an EEL.) Will Asterisk handle that scenario with common drivers and cards? Who generates the customer audio comfort sounds, ringing, busy, etc? Cary I know a lot, but not everything. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
Be careful, telcos may make the users responsible if they have insecure PBXes...right now they often write off much of the charges. But I do agree that there would be a lot less garbage on the net if everyone was liable for their insecurity. Heck, there would be no SIP attacks if everyone's systems were secure - there would be no gain in trying to exploit reasonably unexploitable systems. On Nov 1, 2010, at 11:54 AM, jon pounder j...@inline.net wrote: On 11/01/2010 01:44 PM, Nyamul Hassan wrote: I think the only real solution here is to make people take more responsibility for their actions - find and punish the actual abusers - make users liable for damages caused by infected PC's - defaults from an isp should be everything locked down but with user able to request more ports being opened at no extra cost, if a user asks for it they then take on responsibility for the use of that port. LOL On Mon, Nov 1, 2010 at 23:33, Cary Fitch ca...@usawide.net wrote: I was going to point out a failing of the attackers, but figured they read the list and don’t need any more tips. Cary Fitch From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Monday, November 01, 2010 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: Under heavy attack And obviously these attackers read our emails on lists like this and adjust their sick strategies accordingly. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 12:02 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Only 100? We had a single server over 300. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve international calls anywa... Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net wrote: No. It seems that opening ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
On Sun, Oct 31, 2010 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Sat, Oct 30, 2010 at 07:33:23PM -0600, Joel Maslak wrote: The CPU usage is trivial to deny them. As is the bandwidth usage, if you are not sitting on a slowish broadband connection. s/slow/assymetric/ A 1mb/s uplink is slow nowadays. I suspect a symetrical 1mb/s SDSL line would also be having trouble with lots of registrations. But regardless, that's why I don't use ADSL for call paths, unless the ADSL is 100% within a corporate network (terminates on an ATM line in some corporate office, not in a public provider) - to easy for bad guys to send enough traffic at you to disrupt your calls. If you did have fast enough downlink to not be a victim of this, then you just need QoS - VoIP signalling (registration/registration-fail messages) should always be a lower priority than the VoIP media stream - and it's possible even on ADSL internet connections to control what you send to your provider and in what order you send it. Media packets should always be sent before signaling on that uplink. Even fair queuing (so long as your router recognizes the UDP traffic flows as flows) would help (and would let your legitimate users register quickly even during an attack). It also seems that the only way to make blocking effective is to block everything by default except known endpoints. Blocking the door knickers doesn't protect against a bad guy finding (not through brute force) valid credentials. Unless you have people on the road. Agreed. But I would host that in a datacenter with adequate bandwidth, not on the end of an ADSL or other connection that is easy to DOS. If these are mobile users, I hope they never use any public networks (hotels, starbucks) where other subscribers can do things like ARP attacks to do MITM (and steal your calls; it might not be happening today, but it will be happening soon - as the social networking attacks demonstrate). If you do have truly roaming users, I hope you use HTTPS (with validation of certs turned on) or a VPN (likely not an option of connecting to an ADSL site, due to bandwidth concerns). Or unless you have people who want to actually use the peer-to-peer nature of SIP and call your SIP address. Once again, I'd use a border gateway at a datacenter or other location with significant bandwidth (not an ADSL line). Even for a small shop. I suspect even munin would provide you such options. Not to mention any more capable monitor. I already have a monitor (tied into nagios, which pages me if my fraud thresholds are exceeded), but I feel that is probably beyond the abilities of most of the people experiencing call fraud. The people who know what they are doing with Unix and Asterisk are generally not the victims of this. It would be nice if there was something built into Asterisk to alert on fraud - something that an end user with little Asterisk (or Unix) experience could utilize to be alerted to call fraud, which is easily detectable almost 100% of the time (too many calls for the organization == call fraud). And that is really what this is about - keeping someone from getting a $30,000 phone bill. It certainly should be the part of any commercial offering. I stand by my statements. Blocking people who were already denied access will not stop call fraud on systems with secure authentication. You need to worry about the guy that has the trojan on the computer with the soft phone - the hacker who now has legit credentials (and will never be flagged by fail2ban when he uses them). It's the bad guy you don't know about, not the bad guy you stopped, that is a danger. As for bandwidth issues, I would never use an ADSL-based internet connection for VoIP - too easy for the bad guy to make a mess of things (or even just a misconfigured endpoint). But if I did, I'd agree that some sort of fail2ban-like system would be helpful if you couldn't implement QoS. People can take or leave my advice, but it is sound. Practice security theater or actual security. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
On Oct 31, 2010, at 9:57 AM, Jeff LaCoursiere j...@sunfone.com wrote: This only tells you after it is way too late that you now have upstream bills to wrangle with your carriers about, or (like in my case) that your balance is now depeleted, if it trips anything at all. In my very recent case only FIVE calls, all placed at the same time, caused charges of over US$8K as they stayed connected for over two days. This would not have tripped any erlang threshold, and you don't even know that it is affecting your balance until the calls cease. It would have alerted me within 24 hours, which would have been 1/2 the cost. Of course I have an average erlong much lower than 5 over 24 hours. How did they get in? Did they guess a password to get in? Was the password a good, complex password? Or did they get in a different way? That said (thinking out long), I might need to add a trigger for long-lived calls. Even one long lived call to the wrong destination would cost significant money. Maybe I should notify on any call longer than 3 hours during the day, 2 hours long at night? I'll have to look through my CDRs to see how often this would trigger in my environment. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
On Oct 31, 2010, at 9:39 AM, Mark Deneen mden...@gmail.com wrote: On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslak jmas...@antelope.net wrote: If these are mobile users, I hope they never use any public networks (hotels, starbucks) where other subscribers can do things like ARP attacks to do MITM (and steal your calls; it might not be happening today, but it will be happening soon - as the social networking attacks demonstrate). If you do have truly roaming users, I hope you use HTTPS (with validation of certs turned on) or a VPN (likely not an option of connecting to an ADSL site, due to bandwidth concerns). Can you explain why VPN is not an option for ADSL? (Open)VPN overhead is not that high. ~70 bytes per packet if I remember correctly. I can't remember how big OpenVPN's overhead is, but RTP packets are very small (I want to say a 128 byte payload for G711 codecs and 20ms sample per packet). So that overhead is much more significant than it would be for, say, HTTP. It also increases latency for that packet (longer packets take longer) and often jitter (this is a bit more complex, but basically the shorter all the packets are the more manageable jitter is for QoS). RTP over VPN will have lower quality, assuming you deal with any non-QoS links (such as the internet). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
On Oct 31, 2010, at 9:40 AM, jon pounder j...@inline.net wrote: what are you using that is tied to nagios ? I'll package it up next week and make it available. Basically, I use nrpe to call a shell script that looks at the last five minutes, 60 minutes, and 1440 minutes of a asterisk -rx 'core show channels' output that I run from cron every minute (I count the number of paid channels in use [I ignore channels that have no cost associated with them, such as users calling other users]). If any of these thresholds exceeds my error threshold, I signal a nagios CRITICAL alert. Otherwise I return OK. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
To guess an 8 character (which is short) password that consists of random upper case, lower case, numbers, and 10 symbols (there are more you can use if you want), the average number of passwords that you would have to try to get in is: (72^8) / 2 = 361,102,068,154,368 guesses Over a 10 mb/s ethernet link, assuming no latency, if it takes 100 bytes (it actually takes more), with each byte being 8 bits, of traffic sent by the attacker to Asterisk per password guessed, and the attacker knows you use 8 character passwords, then someone would need to do this for 28,888,165,452 seconds, or over 908 years of continuous guessing while saturating a 10 mb/s ethernet link. If the attacker is unlucky, it might take twice as long. It would be only 9 years if you could fill a 1 gigabit link. If this is too short, add one character (9 total) and it will now take 72 times longer. Two characters, and 5,184 times. (math is: ((361,102,068,154,368 * 100bytes) * 8bits) / 10,000,000 bit/s) = 28,888,165,452 seconds) This assumes the attacker knows the peer name (I'm assuming everyone has set their asterisk to not let the attacker know if an peer name is valid). It's actually quicker to crack modern encryption algorithms than to guess good passwords. If you have passwords that are shorter, contain less characters, or are obvious (such as matching extension numbers), then it could take less time. That's why good passwords are important. Good passwords should be truly random, contain a lot of characters, and include as many different classes of character as possible. If you do easy passwords, you'll probably get hacked with or without blocking attackers, if you allow SIP registrations from the internet. I don't think blocking attackers is bad, just not something that actually improves security against fraud. I don't do it - the risk of blocking legitimate users is too high, but others would make different choices, which is fine. I just think it's a false sense of security if you think it makes a difference in preventing fraud. Good passwords do prevent fraud. Monitoring contains fraud. On Oct 31, 2010, at 10:56 AM, C F shma...@gmail.com wrote: Like I said before RUBBISH. One should just ban/block IPs that are attacking you and not let them connect at all. Not just protect against them with fancy passwords. BTW, even your fancy passwords are breakable, can't wait for the day that you'll wake up and smell the coffee. On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslak jmas...@antelope.net wrote: On Sun, Oct 31, 2010 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Oct 30, 2010 at 07:33:23PM -0600, Joel Maslak wrote: The CPU usage is trivial to deny them. As is the bandwidth usage, if you are not sitting on a slowish broadband connection. s/slow/assymetric/ A 1mb/s uplink is slow nowadays. I suspect a symetrical 1mb/s SDSL line would also be having trouble with lots of registrations. But regardless, that's why I don't use ADSL for call paths, unless the ADSL is 100% within a corporate network (terminates on an ATM line in some corporate office, not in a public provider) - to easy for bad guys to send enough traffic at you to disrupt your calls. RUBBISH RUBBISH RUBBISH and RUBBISH again. If you have someone attacking you just block him. If you did have fast enough downlink to not be a victim of this, then you just need QoS - VoIP signalling (registration/registration-fail messages) should always be a lower priority than the VoIP media stream - and it's possible even on ADSL internet connections to control what you send to your provider and in what order you send it. Media packets should always be sent before signaling on that uplink. Even fair queuing (so long as your router recognizes the UDP traffic flows as flows) would help (and would let your legitimate users register quickly even during an attack). Cute idea and should be done maybe for other reasons but nothing to do with attacks, if you are being attacked block the IP. It also seems that the only way to make blocking effective is to block everything by default except known endpoints. Blocking the door knickers doesn't protect against a bad guy finding (not through brute force) valid credentials. Unless you have people on the road. Agreed. But I would host that in a datacenter with adequate bandwidth, not on the end of an ADSL or other connection that is easy to DOS. Why is a datacenter harder to DOS? The fact that there is more bandwidth doesn't in any way make it harder to DOS. BTW, most datacenter in the US do charge based on 95th% If these are mobile users, I hope they never use any public networks (hotels, starbucks) where other subscribers can do things like ARP attacks to do MITM (and steal your calls; it might not be happening today, but it will be happening soon - as the social networking attacks demonstrate). If you do have
Re: [asterisk-users] Under heavy attack
Is there really any benefit to blocking these, if you use good passwords? On Sat, Oct 30, 2010 at 1:20 PM, Warren Selby wcse...@selbytech.com wrote: I'm experiencing this on one of my clients servers. The attack is ongoing. Thanks, --Warren Selby On Oct 30, 2010, at 2:28 PM, Zeeshan Zakaria zisha...@gmail.com wrote: My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- http://www.ilovetovoip.comwww.ilovetovoip.com http://www.pbxforall.comwww.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is digium doing on port 113?
Probably doing an ident lookup when you send mail to the list. Standard sendmail behavior. On Oct 30, 2010, at 5:37 PM, Hans Witvliet h...@a-domani.nl wrote: While on the subject, what is digium doing on my port 113? just from my logfile: Oct 31 01:11:07 fw2 kernel: EXT; INC, INTRUDER IN=eth0 OUT= MAC=08:00:20:da:3b:4a:00:90:1a:42:70:d3:08:00 SRC=216.207.245.17 LEN=40 TOS=0x00 PREC=0x00 TTL=247 ID=15394 PROTO=TCP SPT=56211 DPT=113 WINDOW=0 RES=0x00 RST URGP=0 host 216.207.245.17 17.245.207.216.in-addr.arpa domain name pointer lists.digium.com. I'm not logged @digium, not compiling, not accessing list archives retieving svn's From http://www.unidata.ucar.edu/support/help/MailArchives/idd/msg00983.html Port 113 supports what is known as an IDENT service. Basically, it tries to determine the remote user of a given client network connection. Yesterday, our web server (128.117.149.62) logged several connections from mail.arilabs.com (206.129.115.118) to which it attempts a connection on port 113. If it is sucessful, it will determine the remote user who connected. This service is widely used on Unix systems, but not really supported on Windows or Mac operating systems. So why is the list-server sending an ident-REQ to my IP? It is blocked anyway, bur WHY??? hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
No. It seems that opening up some sort of automatic blocking could cause an attacker forging packets to block legitimate endpoints. It also seems like they won't get in with good passwords, so it isn't actually accomplishing something to worry about the script kiddies if you have good passwords. And this blocking won't actually stop someone with a zero day attack or who is sophisticated and can attack from many IP addresses - these are the real threats for people with good passwords. The CPU usage is trivial to deny them. As is the bandwidth usage, if you are not sitting on a slowish broadband connection. Sure blocking doesn't hurt, but does the help it provides exceed the downsides (effort and risk of blocking legitimate users)? I suspect it doesn't...if you have strong passwords. If you have weak passwords, you should fix that. It also seems that the only way to make blocking effective is to block everything by default except known endpoints. Blocking the door knickers doesn't protect against a bad guy finding (not through brute force) valid credentials. For me, monitoring outbound call volume makes a lot more sense. I would love to see an easy to use, out of the box method to alert me if more than x number of erlangs* are exceeded within a five minute, sixty minute, and one day time period. For me, I would want alerting on more than 10 erlangs over five minutes, 8 over an hour, and 2 over a day. Exceeding these would likely indicate fraud for my installation. Smaller sites would use smaller numbers, larger ones would use bigger ones. *erlang: one erlang represents full utilization of a single call path over the monitoring period. The monitoring period is usually one hour, but can be anything (5, 60, or 1440 minutes in this case). On Oct 30, 2010, at 6:53 PM, C F shma...@gmail.com wrote: You kidding? On Sat, Oct 30, 2010 at 3:43 PM, Joel Maslak jmas...@antelope.net wrote: Is there really any benefit to blocking these, if you use good passwords? On Sat, Oct 30, 2010 at 1:20 PM, Warren Selby wcse...@selbytech.com wrote: I'm experiencing this on one of my clients servers. The attack is ongoing. Thanks, --Warren Selby On Oct 30, 2010, at 2:28 PM, Zeeshan Zakaria zisha...@gmail.com wrote: My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
Ah, that makes sense - I probably would restrict to only known endpoints by IP address if I has only DSL bandwidth. But blocking attackers makes sense if that isn't an option. Yes, they are after cheap calls. On Oct 30, 2010, at 7:23 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sat, Oct 30, 2010 at 01:43:49PM -0600, Joel Maslak wrote: Is there really any benefit to blocking these, if you use good passwords? Regardless of any threat from those attacks succeeding, they completely saturated the uplink in our ADSL-connected office. What are they after, anyway? Merely cheap international calls? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio problems on cable modem link
On Fri, Oct 15, 2010 at 8:53 AM, Michelle Dupuis mdup...@ocg.ca wrote: When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We're using ulaw, and confirmed that traffic with 2 calls is 175kbps in/out. (IAX connection out) Asterisk doesn't report any dropped frames, the internet connection looks fine, etc. We have a linux router in place running wondershaper that seems to be running fine (same as our other installations). Can someone suggest where to look? Could this be the ITSP? It could be your traffic shapper, the ITSP, your local network, the ISP's network, or the internet backbone - basically anywhere in-between. You only have control over your local network, so I'd start there. Look for duplex mismatches (hint: if one end is set to auto or not able to be set manually, the other end should also be auto, never full [don't worry, they'll negotiate full, but only if both ends are set to auto; otherwise, the auto end will negotiate half due to the end running full not broadcasting capabilities when hardset]). That said, I've never felt great about using the internet for phone calls - you can't controll anything else in the chain, so the possibility of problems is huge - and most of the time you can't fix it. I know lots of people here do it, but it's going to be problematic. If you want toll-quality voice, you still need either TDM lines or dedicated (non-internet) bandwidth. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] drop dead fix
On Fri, Oct 15, 2010 at 9:35 AM, Danny Nicholas da...@debsinc.com wrote: Don't know if this will make acceptable GSM files, but should help with the WAV ones. Are you using GSM to talk to an ITSP (the idea of banking voice calls going across the internet makes me cringe)? If not, what are you using GSM for? GSM always sounds like garbage (and see below - it's not what you are hearing on your mobile phone! It's not as good as mobile phone codecs). If you are using GSM to save bandwidth, you should really look at a better codec - but I would think a banking system wouldn't use the internet for the voice channel. If you are using a private network and bandwidth is still a concern, I'd look at any of the other codecs (except maybe ilbc, which is even worse than GSM). Any of them would sound better. Somehow, to get to a mobile handset user (who uses GSM), the call will hit the PTSN. The PTSN, as others mentioned, is 8K alaw or ulaw (depending on your country). Get the recordings to sound good on the PTSN (convert to alaw or ulaw 8K, as that's what will happen NO MATTER WHAT when your call hits the PTSN) - don't even try to optimize anything else until then. If you're hitting the PTSN at all (versus a direct connection within an IP-based GSM provider's network - unlikely that you have this), even though the handset user is on GSM, you do NOT want to use GSM as your encoding. Use 8K alaw/ulaw (wav format). I suspect your GSM providers in your area have spent literally millions of dollars on their GSM encoding systems - let them do the work. They'll have to do it even if you played the GSM file, it'll just sound worse if you play GSM, convert it to 8K alaw/ulaw over the PTSN, then have it converted using a different algorithm at the cell site. Finally, not all mobile calls on even GSM networks are gsm format. If they are a different format, converting from one compressed algorithm (gsm) to another (whatever the carrier uses) is going to sound horrible. So don't bother with the GSM format. Few people you are calling/called-by use that codec (not even the mobile phone users). It'll get resampled into something else. You'd be better off using the raw, basically-uncompressed (I know, I know, not quite accurate) alaw/ulaw - which everyone's codecs are designed to handle very well (since every single PTSN call uses it). For reference, Asterisk uses (I believe) the full rate GSM codec. Mobile phones on most GSM networks are using an AMR (not full rate) codec, as it simply sounds better, can deal with bad connections better, and can even use less bandwidth. Of course it is licensed and patented, so Asterisk doesn't implement it. But because of this, Asterisk's gsm doesn't sound as good as a call on a GSM network. Why would you want that? Just don't use it! See http://en.wikipedia.org/wiki/Adaptive_Multi-Rate (What mobile companies use) And http://en.wikipedia.org/wiki/Full_Rate (What Asterisk uses) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 check with nagios, how to?
Enjoy...you can ignore certain T1/E1 ports if you pass in the name of the
port as an argument (I use this on ports that aren't yet connected to a
telco, but I don't want to get an alert on). I execute it via NRPE on the
Asterisk box. It will give you descriptions of which ports are bad, so you
don't need to guess. :)
#!/usr/bin/perl -w
#
# Copyright (C) 2010 Local Matters, Inc.
# http://www.localmatters.com/
# Author: Joel C. Maslak
#
# Licensed under GPL version 3
#
use strict;
use Carp;
my %ignore;
MAIN: {
my @out = `/usr/sbin/dahdi_scan`;
for my $ig (@ARGV) {
$ignore{$ig} = 1;
}
my $alarm;
my $desc;
my @alarms;
for my $line (@out) {
chomp($line);
if ($line =~ /^alarms=/) {
$alarm = $line;
$alarm =~ s/^alarms=//;
}
if ($line =~ /^description=/) {
$desc = $line;
$desc =~ s/^description=//;
if (!defined($ignore{$desc})) {
if ($alarm ne 'OK') {
push @alarms, $desc: $alarm Alarm;
}
}
}
}
if (scalar(@alarms) 0) {
my $out = join '; ', @alarms;
print Circuits in alarm: $out\n;
exit(2);
} else {
print All monitored circuits OK\n;
exit(0);
}
}
On Tue, Sep 28, 2010 at 9:17 AM, Mark Deneen mden...@gmail.com wrote:
Are you monitoring some dahdi hardware or a separate black box?
If dahdi, you could write a nagios plugin in shell with something like
this:
ALARMS=`dahdi_scan | grep alarms | grep -v OK | wc -l`
and then set the appropriate exit code if ALARMS is not 0.
-M
On Tue, Sep 28, 2010 at 9:22 AM, Dario Quiroz darioqui...@gmail.com
wrote:
We need to monitorate the E1 with nagios, somebody did this? any ideia?
Thanks in advance!
--
Atenciosamente,
---
Dario Quiroz
(71) 9275-9080
gtalk: darioqui...@gmail.com
---
--
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[asterisk-users] Rotary phone on Asterisk
I'm trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial). These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino channel bank which is connected to Asterisk via a Sangnoma card (T1 with echo cancellation). Other phones (touch tone) work fine, as does any phone with a pulse/tone switch, even when these electronic phones are in pulse mode. I'm thinking that Asterisk is a bit too picky about the timing of the rotary dial pulses to handle a mechanical system. Is there any way to correct this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rotary phone on Asterisk
My understanding was that pulse dialing from a channel bank was iffy, but not pulse reception, so long as the channel bank properly reports on/off hook state - that there is no real pulse detection in the channel bank, simply on/off hook status (looking at some of my documentation, real D-2, D-3, and D-4 channel banks all used the LSB in the 6th code word to indicate on/off hook status). Is this no longer correct? I'm using ESF, although I think D4 would work similarly for this function. I will contact Rhino as well, though - just to cover my bases. I was very impressed with their support previously. On Fri, Sep 17, 2010 at 12:56 PM, John Novack jnov...@stromberg-carlson.org wrote: Danny Nicholas wrote: -- *From:* asterisk-users-boun...@lists.digium.com [ mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com] *On Behalf Of *Joel Maslak *Sent:* Friday, September 17, 2010 12:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Rotary phone on Asterisk I'm trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial). These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino channel bank which is connected to Asterisk via a Sangnoma card (T1 with echo cancellation). Other phones (touch tone) work fine, as does any phone with a pulse/tone switch, even when these electronic phones are in pulse mode. I'm thinking that Asterisk is a bit too picky about the timing of the rotary dial pulses to handle a mechanical system. Is there any way to correct this? Check this out http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing Better read the link That refers to the TDM400 card. Posted to voip-info quite a few years ago, I believe it MIGHT have made it into a recent release, or coming soon. For T1 though, I believe the channel bank might be the issue. We have several channel banks of various types with various T1 cards. Adtran 750's aren't supposed to work properly, according to Adtran, other companies do. Better ask Rhino first, as the pulse detection and timing is a channel bank issue. 500 sets are pretty good with pulse speed and make/break ratio. The #9 dials are even better than the #7 ones, though all are fairly stable. Older phones, WE or others, can be more difficult, and may need to be repaired. John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving from DSL to T1
On Mon, Sep 13, 2010 at 1:32 AM, Kevin Keane subscript...@kkeane.comwrote: My numbers are from an ATT DSL line in California, suburban San Diego county, and just around the corner from the central office. So it is not the distance (with DSL, the distance does make quite a difference). On the other hand, there are several hops just to get to the Internet backbone. Lots of misconceptions in this thread. I'll limit this discussion to what I know - ADSL, T1, and cable as delivered by US telcos/ISPs. I just measured my Denver, CO Qwest DSL line. I have a short DSL line - about 4 city blocks, but then, like most metro areas served by Qwest, once my connection ends up in the exchange, I'm backhauled about 30 miles via ATM to the actual edge router(s) that serve the metro area (the edge router is not in the central office).: 10 packets transmitted, 10 packets received, 0.0% packet loss round-trip min/avg/max/stddev = 28.646/31.359/33.828/1.603 ms This is pretty typical of Qwest DSL in Colorado and Wyoming. I've literally run hundreds of DSL sites in these states, and find that the DSL product is a great bargain for business (I usually didn't run it to Qwest's edge router but to my own corporate routers, as basically an ATM circuit that just happens to be delivered on DSL). I ran a large video conferencing network on it without any issues (H323). The reason for the times being around 30 ms round trip is that Qwest uses interleaving on the DSL circuit. It's on for a reason (it makes TCP streams faster - instead of retransmitting whole 1500 byte TCP packets, the system retransmits the 53 byte ATM cell instead. It might have to do that several times, for several packets that get corrupted, but it will still be faster than TCP dealing with the loss. Of course VoIP doesn't use TCP, nor does it need 100% guaranteed packet delivery. But this also can help with jitter, depending on what else is on the line. Typical T1s do not do this interleaving (they are better engineered, regardless of whether they are delivered on true T1 media or backhauled via HDSL - either way, the packets make it out the other end a lot more reliably than they do on ADSL connections). For DSL, there are no shared connections between the telephone exchange and the home - everything there is dedicated. Cable shares a connection with your neighbors. T1 is dedicated to the Exchange as well (burstable or dedicated bandwidth both). At the Exchange, DSL is aggregated on a large (OC3 typically) ATM circuit to get transported to the ISP (or telco's own) edge router. This is shared bandwidth with everyone else on the same DSLAM. A point-to-point T1 has dedicated bandwidth (no aggregation) to the ISP. A frame relay, MPLS, or ATM T1 has the amount of dedicated bandwidth you pay for (in my experience, usually none, as that's cheapest) with the rest of the bandwidth provided if there is capacity on that ATM circuit between the central office and the ISP/telco edge router. Cable is also usually aggregated at some intermediate point, similar to DSL. Once connected to the ISP edge router, all circuits are aggregated out to the internet backbone via a connection smaller than the sum of all the subscriber bandwidths. So, frame relay, MPLS, and ATM T1s - as typically ordered - often function like DSL, with the same choke points. Even a point-to-point T1 that goes to the internet however will hit a choke point and might have packet loss - no matter how much you paid for the T1 (the internet backbone itself has packet loss and choke points). Cable has an additional choke point (subscriber loop). Now, hopefully the ISP and telco have engineered everything to not have significant choke points and you'll never have a capacity problem (the same goes with their peering connections - hopefully they, too, are big enough). But even an MPLS burstable T1 could perform badly at high capacity times. Most of these choke points (such as the DSL DSLAM to edge router) do know to not let one user monopolize the uplink, but to let each user have the same approximate capacity when there is congestion. So someone with say only one VoIP call won't experience packet loss or changes, while another user downloading a movie will see a slower download. (in the IP world, this is called fair queuing - in the ATM world, it goes by other names; they idea is that you should give everyone the same amount of possible bandwidth in a congestion situation, even if they aren't using all of that bandwidth). Most of these technologies do not let you apply QoS to the choke points effectively. So you are left with point-to-point T1s or other T1s that you buy with guaranteed (more expensive) bandwidth. But you can't QoS across the internet. Sure, you can do some traffic shaping on a DSL line, and it'll work good most of the time, but there are no guarantees with guaranteed bandwidth! So, the only way to gaurantee packet delivery is to build your voice IP network like you
Re: [asterisk-users] Moving from DSL to T1
On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl wrote: No these are also geo-stationary (same altitude, so same delay), commercial and military satelites, Yes, exactly. Geostationary satellites have been used for telephone for ages (and are still used for remote areas - they have advantages over the disintegrating constellations such as iridium - namely predictability). As for consumer (home) grade satellite internet service, it's pretty low quality. But if you have money, you can have just as good of service as the telcos enjoy for TDM voice over them (even with VoIP). I know several organizations using them (but they are paying more than the $100 or so a month as is typical for a home user - a lot more). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom dhcp boot
Use lowercase for ftp:// . That might be the issue but it should be easy to test. Do your FTP server logs shpw anything? On Sep 10, 2010, at 5:35 PM, colin mcdermott colinjamesmcderm...@gmail.com wrote: Hi all I have a few Polycom 331's but after following allot of advice I can't get them to provision from a dhcp boot server. We have a sonicwall router in place. I can press setup and set the FTP boot server to my * box. From there th phones boot fine. But I cannot get them to autoprovision. I have tried dhcp option 66=ipaddres. 66=FTP://PlcmSpIp:plcms...@192.168.1.1/ u ahve also tried options 129, 150, 160, etc. I realise that this is not an asterisk issue. But does anyone have any experience on this (particularly using sonicwall routers for Dhcp)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangnoma + Digium Bridging
I'm trying to install both a Sangnoma A102 (with echo cancellation) card and a Digium 8 port analog card with echo cancellation (Digium AEX800E) in the same server. I know I probably shouldn't have mixed vendors - lesson learned for next time. That said, I have everything working fine...except Native Bridging between the Sangnoma and Digium cards. When I do native bridging, I get a very distorted sound (unusable, not just bad sounding). Is it possible to do native bridging between them? I am using a workaround through a local channel now, so I have things working good enough for my purposes, but I'd like to cut out the local channel if possible. I'm running Asterisk 1.6.2.11 / DAHDI 2.3.0.1 / Wanpipe 3.4.9 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangnoma + Digium Bridging
In a moment of inspiration, I recompiled both DAHDI and Wanpipe - and this seemed to have resolved my issues, all is working great now. On Wed, Sep 8, 2010 at 10:52 AM, Joel Maslak jmas...@antelope.net wrote: I'm trying to install both a Sangnoma A102 (with echo cancellation) card and a Digium 8 port analog card with echo cancellation (Digium AEX800E) in the same server. I know I probably shouldn't have mixed vendors - lesson learned for next time. That said, I have everything working fine...except Native Bridging between the Sangnoma and Digium cards. When I do native bridging, I get a very distorted sound (unusable, not just bad sounding). Is it possible to do native bridging between them? I am using a workaround through a local channel now, so I have things working good enough for my purposes, but I'd like to cut out the local channel if possible. I'm running Asterisk 1.6.2.11 / DAHDI 2.3.0.1 / Wanpipe 3.4.9 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes
g711 across a network without perfect jitter/delay characteristics will not work. You cannot do g711 faxing across the internet - at all. It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes). On Fri, Sep 3, 2010 at 12:32 PM, dave george dgeo...@teletoneinc.comwrote: Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of Storage Area Network with Asterisk
On Sun, Aug 15, 2010 at 9:09 AM, Michelle Dupuis mdup...@ocg.ca wrote: Are there any best practices for using a SAN with Asterisk? In the past we've kept config files local, but voicemail on a SAN. Aree there any issues with latency putting voice prompts, configs, etc. on a SAN? Anyone have some best practices to share? We mount up a Netapp SAN for backup purposes. We rsync the live files (/etc, /var/spool/asterisk) to the SAN hourly for backup (losing an hour of voicemail wouldn't hurt us that much), but you could rsync at a different frequency. But all live files Asterisk uses, including voice prompts, are served out of the local file system on top of RAID-1 local disk. We did this to allow Asterisk to continue functioning in he midst of a SAN/network outage - backups will error out or hang, but Asterisk will keep going. We push out voice prompts and most config files via Puppet ( http://www.puppetlabs.com/) - with the Puppet repository being backed by an SVN repository so we have version control of all the changes we push out. We do this for other systems (such as web servers) to ensure all the systems end up with the same versions of files as each other. The only downside is they don't all get the changes at exactly the same time, but for something like voice prompts and configs I would think that won't matter (voicemail is a different beast). As for voicemail, if I was running redundant voicemail servers, I'd probably do things differently - put the voicemails on a SAN of some kind, perhaps even modifying Asterisk (with the voice mail left hook) to copy any new voicemail to the other box after it is left, if the other box is responsive. Then, I would write something that could merge two voicemail stores (message 1 on VM store 1 might not be the same as message 1 on VM store 2 - if not, copy it over as a new message, not overwriting the old one). My principle with this has been Don't make Asterisk depend on anything it doesn't absolutely have to depend upon. But I do think you could run prompts and configs off of a SAN - no problem there - but just that you would be building a dependency that would cause Asterisk to have issues if the SAN went offline or became unreachable. How reliable is your network/SAN? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax/Modem, Asterisk, Channel Banks
I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI T1. In the other port, I'd terminate a Rhino channel bank, connected to each of my fax machines (and a stamp machine with an internal modem). What I'm wanting is to be able to send/receive faxes via the telco PRI and the analog fax machines. I also want the stamp machine to work. I don't want this to work 98% as well as the Telco - they truly need to work 100% as well. So...will this work? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
On Tue, Aug 3, 2010 at 1:30 PM, Mark G. Thomas m...@misty.com wrote: I didn't know there was a U option. I don't see any mention of it on the voip-info.org wiki or other Dial() documentation, but didn't check for new options in the built in documentation until just now. I updated the dial documentation on voip-info.org - but I'm sure I didn't do it perfectly. I also re-ordered the options by ASCII sort order, rather than the random order (I think) that was there before. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integration with Toshiba Strata DK424
to the Toshiba system, which is best done with a blind transfer on the Toshiba side - something I'd have trouble doing if I sent it over the fake E1 span. Example: exten = 850,1,Flash() exten = 850,n,SendDTMF(850) exten = 850,n,Hangup() The Toshiba MWI lights are set via an external notify script from voicemail, that initiates a call with the proper codes. Things I'd like, that I don't have, with this current integration are: 1) Knowing who is calling into the Toshiba, when the call is forwarded by the Toshiba. Basically, caller ID. 2) Complete elimination of the A tone, which doesn't seem possible if the D tone is still sent. I'd be interested in hearing how others may have done this, if you have experience with this. OT: The company I work for is hiring! We're hiring a Senior System Administration and several software engineering positions in Denver. Visit: http://www.localmatters.com/careers -- Joel Maslak -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comments on Voice Quality IP Hard Phones
On Wed, 25 Feb 2004, Peer Oliver schmidt wrote: Budgetone I have seen a few reviews, but none go to deep into the voice quality issue. I don't mind the voice quality, I just wish it would always be working when I picked up the handset. Mine tend to lock up (they are behind a firewall, I am running current CVS, have upgraded firmware, etc). If they fixed their software (maybe even opened it up for others to write), I think it would be a heck of a deal for a phone. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference and transfer
On Thu, 26 Feb 2004, Rana Dutt wrote: Also, I find it disconcerting that there's a Conference button on the Grandstream phone, but when it's pressed, nothing happens. If this sends out some sort of switch-hook flash, can Asterisk intercept it, and then use the meetme app? Do the Cisco phones support conferencing using the Conference button without the need for the meetme app? My understanding is that the purpose of the button is to look pretty unless you have the higher-end budget-tone (102?) where it then does 3 way calling. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Woodpeckers
On Thu, 19 Feb 2004, Nick Bachmann wrote: 300Hz is pretty high to filter out... it's still well within the rage of voices. To compare, 300Hz is about a diatonic concert D. POTS has been filtering out 300HZ and 3000HZ for years. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing 800 numbers with VOIP
On Tue, 10 Feb 2004, Tim Petlock wrote: Hm. After seeing all the people who say it works, I thought - maybe I forgot to dial 9 in front of the number and that's why the call failed. So I looked up the Wells Fargo toll free number again and tried it. Failed. SIT tones and We're sorry, your call did not go through. Will you please try your call again later? The recording has nothing at the end that might give some clue who was generating it either. I just tested Vonage and VoicePulse - both worked fine. You might want to check to verify that you don't have 1-800 numbers going out a different service then you expect. IIRC, when you set up IAXTEL, it recommends you add lines to handle 1-800 numbers. So if you do that, and your IAXTEL connection stops working, you'll lose the ability to dial 1-800 numbers. I don't have a Nufone account (Jeremy - if you are reading - I would probably have one if there was a price for a starter package listed on your site - something for SoHo use, without any deep discounts or anything, just something to use to play with the service; I have a personal aversion to bothering with companies who don't list their prices), so I have no idea if Nufone's 1-800 service works or not. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring Firefly Network in *
I did get it to work, and can place and receive calls through the Firefly
network via *.
Compared to iaxtel or FWD, there is a significantly higher amount of
latency, but it is workable.
For some reason, this needed to be the last entry in my iax.conf or it
would try to authenticate with a different user ID when receiving calls
(and obviously would fail.
Relevant section from my iax.conf:
register = 87210384:[EMAIL PROTECTED]
[87210384]
context = firefly-in
secret=xxx
auth=md5
type=friend
username=87210384
host=firefly.virbiage.com
qualify=yes
trunk=no
In my dialplan:
[globals]
...
FIREFLY=IAX2/[EMAIL PROTECTED]
[fireflycalls]
; Firefly Calls
;
exten = _**.,1,SetCallerID(87210384)
exten = _**.,2,SetCIDName(Joel Maslak)
exten = _**.,3,Dial(${FIREFLY}/${EXTEN:2},,)
exten = _**.,4,Playback(invalid)
[good-user]
include = extensions
include = extensions-services
include = good-outbound
include = fireflycalls
For inbound:
[firefly-in]
exten = s,1,GotoIfTime(07:00-21:00,mon-fri,*,*?incoming,s,1)
exten = s,2,GotoIfTime(09:30-21:00,*,*,*?incoming,s,1)
exten = s,3,VoiceMail(u10)
Where incoming is my standard incoming context. Calls come in without
any DNID, they go straight into the s extension in the firefly-in
context. The time restrictions are my standard non-PSTN annoyance at
2:00 AM filter.
With this config, I can dial Firefly users by dialing ** + their Firefly
number.
--
Joel Maslak
___
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RE: [Asterisk-Users] [ot] Grandstream hardware
On Sun, 18 Jan 2004 [EMAIL PROTECTED] wrote: On another note, what's the deal with Hold, Mute, Caller ID Review, and Called Number Review on these things? Do they just not work, or am I missing something? Caller ID Review/Called Number Review only works when the phone is off the hook. I don't know why they did it that way. I gave up on trying to transfer with the phone. Transfer doesn't. It just puts the call into never-never-land, which means you can't get back to it but it won't hang up either. Mute works for me, as does hold. And this doesn't even mention the phone forgetting how buttons work (happens every few days on one of my phones - you have to reboot the phone). I won't even mention what I think of the early dial almost working. Or the fact that the handset isn't wired in a standard way (it does not work with some assistive technology because of that). Or that you don't hear hardly any of your own voice in the handset. I don't really have a problem with the phone's hardware (well, other then the way the handset is wired). I have a tremendous problem with its firmware, though. Right now, these phones have absolutely no place in a business at all - not even as reception area phones. I hope they fix this problem, though - they could be decent phones I think for the cost if the firmware got fixed. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static Noise coming from Wildcard FXS: Wildcard TDM400P
On Wed, 14 Jan 2004 [EMAIL PROTECTED] wrote: I recently plugged in Phone to my TDM400P Card to test out something I mostly use sip phones to interface with *. All of sudden I'm getting lot of line static noise coming of the card is there any settings I should look at or anything I need to do on the command line at this point I'm open to any ideas If it is an old card (without the 12V drive power cable connector), stop *, unload the modules, reload them, restart *. That will fix it. You are probably sharing an interrupt with the card and something else in your machine. Get it onto its own interrupt. That fixed the problem for me. My guess is that the driver isn't smart enough to notice this failure and reinitialize the card. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to Order Disconnect Supervision from SBC using Adit 600?
On Tue, 13 Jan 2004, Jonathan Moore wrote: LSRB = Loop Start with Reverse Battery I believe I currently have the lines set to LSCPD which improved the hangup situation, but hasn't completely fixed it. Try LSRB - it may work. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and lawsuits
On Tue, 6 Jan 2004, Jon Pounder wrote: The phone does not have to necessarily be at the pbx either, it could be brought out to the reception desk etc. On Definity systems, we used a device called something like Emergency Cut-over. When power from the switch was lost, the device threw a bunch of relays cutting CO lines over to fax machines that were specifically chosen to allow dialing without power (many fax machines won't dial unless there is power) or to fax machines with a Y adapter connected to a $9 Wal-Mart phone. Normally, these fax machines would go through the switch, but if the switch had problems, it would cut over. We put big signs above all the fax machines indicating that they were EMERGENCY PHONES. I've also seen pay-phones installed in some areas to serve this function as well (typically shop environments where personal phone calls using company equipment were frowned upon). -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encryption
On Thu, 25 Dec 2003, Michael Sandee wrote: Most block and stream ciphers can recover from loss... And if you don't use block chaining, you have no loss at all... (but it's less secure, however this is always relative to what you try to protect) (With block chaining, you lose the next block) (Afaik block ciphers are more secure, but well... i'm no cryptographer, nor cryptanalyst... or mathematician... or ...) It opens up a *LOT* of attacks, most significantly data injection, if you aren't really careful. Here's how I would write the protocol... You have a bunch of input packets which consist of some sort of UDP payload that represents the voice on the channel. These are D (for Data) Each connection has a unique ID. This ID should be based on time and be unique. An example would be Time of connection origination in seconds since 1970 concatinated with a random number. So this number would probably be a 64 bit number - the first four bytes would be the seconds since 1970, the next four bytes would be random. This is CID. Each packet would then have a sequence number - not for the traditional reassmbly reasons, though, simply to keep old packets from being reinjected. This probably isn't necessarily though if * uses sequence numbers in IAX currently. I'm assuming that it is already in the data stream now. To encrypt, you would take a block cipher. The result packet would be (. is string encapsulation, E() is encrypt): P = CID . E(CID . D) To decrypt you would split the packet into CID and C (Ciphertext). And then you would (E'() is decrypt): CID . D = E'(C) You would compare the plain text CID with the value in the encrypted packet, logging an attack if they don't match. You would also need to throw out old CIDs when they tried to initiate a connection. And also old packet sequence numbers. To prevent replay attacks. This would let the packets arrive out of sequence, handle a missing packet fine, and also ensure that two packets containing exactly the same data did not have the same ciphertext (as they would have different sequence numbers and CIDs). These are just some of my thoughts, please don't pin yourself by just looking at the best cipher... which is considered to be AES-CTR (block) by many people... as you see it has 120bits of overhead when used with GSM... (If the 33 bytes figure presented before is right) Cipher capabilities could be exchanged just like codec capabilities... and if a device (IAXy?) only happens to support plain AES... so be it, but please don't restrict the protocol to that :) It would be nice, though, if the cipher something like AES, though, since that would meet government requirements for encryption in the US. It might give us more users. 3DES would work - right now - but not in a few years as 3DES is in the process of being phased out. I do understand the overhead issues, though. I would say that I would have uses for this technology tomorrow if it was cheap and affordable, had a well-engineered protocol (which mine probably isn't - I just threw it out to show one way of trying to solve this problem), etc. I work with some organizations that really do need encrypted voice but can't afford commercial encrypted telephones. It would also be a good way for me to get VoIP into those organizations. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Tue, 23 Dec 2003, Rich Adamson wrote: If a collision or dropped packet occurs (in a voip udp environment) there is no way to retransmit the missing/damaged packet. Missing one packet isn't a big deal, but if you have collisions and/or dropped packets, there is a very high probability that lots of packets will be dropped. If too many are dropped, you'll hear the result in the undecoded voice as choppy voice. Actually, collisions occur at Layer 2, not Layer 3, and the layer 2 hardware automatically resends packets involved in a collision - layer 3 is never aware of it happening (although it may cause additional delay). Eventually the ethernet card will give up if too many collisions occur during retries, but this is very rare in practice unless the network is *VERY* loaded. Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg ethernet would handle roughly 20-25 rtp sessions before bumping into the problem (your milage may vary). The majority of the folks on this list seem to be running home/soho systems and would likely never run into the issue. But the heavier users will. For a duplex mismatch, my experience is that if one end on a 100 Mb/sec link is half and the other is full, bandwidth is limited to about 8 Mb/sec max. This is based on some tests I've accidentally conducted. If you try to send 9 Mb/sec over that link, yes, some packets will get dropped as they simply won't fit. (But I do agree that for a half-half link, you can get about 20 Mb/sec) -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ToIP (TDD over IP)
On Mon, 22 Dec 2003, Philipp von Klitzing wrote: Excuse my ignorance: What exactly is TDD? Is it US specific? TDD - Telecommunications Device for the Deaf (also used by people with speech problems). Also known as a TTY (Telephone Typewriter) or TDY (not sure what it means) I don't know if it is US-specific or not. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ToIP (TDD over IP)
I didn't know if it would work or not, but I figured I'd try slow-speed half-duplex TDD over GSM Vonage. I called a AGI script I have that speaks to TTYs, by calling from Vonage to one of my Voicepulse lines. I don't control the Vonage codec, so I have no idea what it uses, but I am using GSM for the Voicepulse line. Everything worked fine - echo canceling didn't cause any trouble (I don't know if it would have if I did full duplex, though), I didn't lose any characters, etc. So for people who have a need for this kind of technology, I can tell you that it will work. I'm also curious if anyone else is doing this or if anyone else is using the Asterisk TDD support. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P + TDM400P
I thought I'd share my Asterisk experience, which hasn't exactly been as pleasant as I would like but now seems usable in most ways and more then I expected in other ways. I wanted a home PBX system, that would let me treat different callers different ways depending on CID. I initially bought the Digium developer's kit to try things out. That's a single port TDM400 and a X101P. I've added another X101P. One X101P terminates in a Vontage Cisco ATA-186. The other terminates with Qwest. The TDM400 is connected to both a TDD and a cordless phone. I also have a softphone connected along with 2 DID numbers through Voicepulse. I have a second Asterisk system outside my firewall to use for FWD. What went bad (most are minor): - SIP NATing. I just gave up on this. That's why I set up the second * box outside my firewall, with an IAX2 connection from my inside * box to it. - Voicemail - there are lots of little things missing. The main stuff is there, but lots of things I'd expect in a fully functional VM aren't there. I'm also disappointed that there is no way to turn on/off the MWI on a phone except through receiving a VM message (I'm working on a patch to add some basic MWI functionality for things like external VM systems) - TDM400 lockups - Sometimes, the TDM400 card seems to go into crazy static mode. It's the newest revision (this apparently is a known bug in some of the older versions). This hasn't happened since I moved cards around (see next item) and have updated the Asterisk software. The only solution was to unload and reload the wcfxs module. If it comes back, I suspect Digium will stand behind their product (I haven't contacted them formally, so if anything this is half my fault), and it may have been related to the IRQ problems. - IRQ/PCI problems. I have a lot of stuff in this machine that takes IRQs, including SCSI, sound card, net cards, etc. I have 6 filled slots right now. Initially, when I added the second X101P, the machine would not boot. It would either hang when the wcfxo module was loaded or crash with some weird SCSI IRQ errors. After Googling a big I found that some people can correct this by moving the cards around, so I did this. I also disabled serial and parallel ports in my BIOS. It took several tries at moving the cars around before I found a combination that works but things do seem to work right now even with the TDM400 card sharing an interrupt with the USB-UHCI device. - ECHO!!! I ended up updating the * code from CVS and turning on Mark2 with aggressive suppression. This fixed it, although things still sound a bit strange with full duplex talking (the echo suppression doesn't seem to like that and the voice volume changes along with some echo being present). I didn't have much of an echo problem with the X100 going into the ATA-186, but the X100 going to Qwest was miserable. Of course I also have DSL on the line, and the wiring in my house was done by the previous owner (who fancied himself as an electrician), so I'm not claiming my wiring is bad. Of course I never had any echo on my normal phones, nor did my DSL have any problems, so I do think there is something up with the X100 cards. Right now, echo seems okay. - Volume levels - I had to bump up the rxgain on the Qwest circuit a bit. But now all seems well. - Hangup and MWI clearing FSK tones - when hangup is executed after an extension dials out, * sends the FSK tones to clear the MWI (if appropriate). Unfortunately, while * may be hungup, I am not. WHAT WENT RIGHT: - IAX2 - this is slick. Works great between my inside-the-firewall * server and my outside-the-firewall * server, as well as between the inside box and Voicepulse. It would be nice if there was a *tad* better logging by default on incoming IAX calls (I had some problems initially with not having an entry for the Voicepulse DID lines, so * couldn't find the extension Voicepulse was looking for; rather then logging, it just hangs up; The IAX debug logs don't indicate *WHICH* extension is being looked for, either). - IVR functionality - that works great, too. No gripes at all - this is better then what you get with most PBX's - Unexpected functionality - I didn't know my cordless phone had a MWI. But it does. I was very surprised when a light I had never noticed on the phone before lit up after I received a voicemail message. That's a neat feature, and even neater that I didn't need to configure ANYTHING to get that to work. - Preliminary TDD support - this also pleased me, although there are some bugs in this support. It's nice to be able to set up a TDD interactive response system (right now, I can call my home machine, enter an IP address, and see my network management view of that machine through my TDD - which is also nice because lots of places have TDDs connected to pay phones...) [of course it would be nice if...* had a TDD extension that worked like the FAX extension if it heard TDD tones - especially
Re: [Asterisk-Users] 911 settings.
On Fri, 19 Dec 2003, Nick Bachmann wrote: I don't know how big of a customer you are for your phone company, but if you have more than a token number of lines they'll hopefully go for it. Another option is to call the non-emergency number of the dispatch center and explain this one number/address could actually mean someone is calling from either this location or the one down the street... Make sure you get this information from the caller.. Typically they can add some comments to their database at the dispatch center (they typically use this feature for making note of things like site stores 3 million gallons of highly explosive substance, which the phone company doesn't keep in their databases). It's not as good as knowing exactly where the call is coming from, but it is a start. It might be good for people that have non-local phone numbers, too, and 911 is translated to the non-emergency phone number. If you call them up and talk to them, they are typically glad to help. They often are willing to help you test the actual 911 part of your dial-plan, too (are you *SURE* you haven't screwed that up? The only way to find out is to test this - WITH THE DISPATCH CENTER'S COOPERATION). (this isn't a bad thing to do even if you are using a land line with supposedly the right address - all computer databases are not 100% accurate...) -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe: Zap channels don't ever disconnect. . .
On Mon, 15 Dec 2003, Brian Capouch wrote: Anyone know of a way of doing this when the scumbag ILEC won't give you supervision? Probably not much. Try turning on callprogress and/or busydetect - it MIGHT help. But the only way to do this right is with supervision on an analog trunk or with digital circuits. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXtel down?
On Fri, 28 Nov 2003, Steve Rodgers wrote: Anyone else having timeout problems with IAXtel? Here's the logfile output, user names, passwords, and destination phone numbers have been changed to protect the guilty I just called myself. It worked fine. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
On Sat, 22 Nov 2003, PBX wrote: But the problem with that is the user then hears dial tone. And if they hang up the line it rings them back... The only way I have been able to get anything like what I want is, to push flash then the hold button... That is not the exact motion I want a user to go through. If there is a way to signal the hold button in ADSI that would work.. Cause then I could request a flash then hold sequence and then to take them off hold, I could just do the opposite sequence. Try call parking... -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
