Re: [Asterisk-Users] Calls forwarding to numbers only in user's context
Bartosz Piec wrote: How to set calls forwarding only to numbers that are available in user's context (so if he has only locals calls he cannot set calls forwarding for mobile phones)? When the user sets the forwarding number, store the user's context in the DB along with the forwarding number. Make sure you have an invalid extension 'i' in the user's context as well. For an incoming call, when the Dial returns busy or no answer, execute a jump to the retrieved context/extension. If the user doesn't have the forwarding number in his context, the incoming call will land in the invalid extension. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud
Gary Richardson wrote: I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811.. For me, all incoming/outgoing calls arrive/leave via IAX2/ilbc and all the local end points are SIP/ulaw (SPA-841s.) Still haven't seen any recordings cut short on 1.2.4. By now if I were using 1.2.1, I would have seen it at least once or twice. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud
Gary Richardson wrote: Now it seems that if I'm really loud on a call, MixMonitor stops recording. The wav file stops growing. The log says nothing. When you hang up the call, MixMonitor reports that it is exiting, even though it hasn't been recording since that loud noise. Has anyone experienced such a problem with MixMonitor? Is MixMonitor well tested? I've seen exactly this with MixMonitor in 1.2.1, but I hadn't isolated it to volume issues, just random occurrences. I haven't seen it yet in a week on 1.2.4, but I don't know if the bug is gone or it just hasn't triggered yet. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheapest provider for Philippine route
Sam Tam wrote: Do anyone know who can provide some cheap PH routes/.’ I've been looking myself. Cheapest DIDs in Metro Manila I've seen are $27.50/month; cheapest termination to same (non-mobile) from US I've seen is $0.23/minute. Expensive chismis :-) -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Question
Michael Collins wrote: I had never sat down to write an AGI script before - I hadn't needed one - but I thought, How hard can it be? Ugh. The Asterisk::AGI module is very handy, and I highly recommend it. I've only written one AGI script in my life (up to now) but I've written 10's of thousands of lines of Perl and I know a good module when I see one. I'll second that. I just wrote my first AGI script last night--using Python and the Pyst AGI python library. Works wonderfully. Even with AEL, things get cumbersome, and having the facilities of a true language really simplify things. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] courtesy message calling mobile phones
C F wrote: Can you explain this? What country? In this case it's not asterisk but the telco that has to do the Answer. To every mobile? or just that provider? I too have seen something similar in the past. When calling Verizon (408-489) numbers, when there is no answer and it rolls over to voicemail, the callee's greeting plays with no answer indication from Verizon. Eventually the Dial times out while in the middle of the callee's greeting and the caller is not able to leave a voicemail. I don't know if this is still happening now or if it was a just a temporary fluke when it was reported to me. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with voicemail
Roger Lewau wrote: If the voicemailbox contains messages the voicemail application exits with a non-zero status either when reading the number of messages or when selecting 1 for listening to new messages. Is it possible the permissions for the sounds directory or individual files within have changed such that the user asterisk runs under no longer has read access? -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context being ignored by inbound sip call
btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] is there a web interface to this mailing list?
Douglas Garstang wrote: Yes, programming the dialplan is akin to programming assembler. Too funny. But true. The first time I did a 'show dialplan' after trying out AEL, I felt like I was seeing an assembler dump of C++ :-) -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *
Andres wrote: There is no username in the above To header. Check your DIAL command because something is wrong here. Thats why you get a 404. The SPA can't match the username. Yes. I had not reverted to an early enough commit on the configuration files and the usernames were still missing in sip.conf. Thanks. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *
I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server was working. Doesn't fix it. I reset one of the SPA841s to factory defaults and reconfigured, still has the problem. Outbound calls from the SPA841s through the * server work fine. How do I figure out what the SPAs are unhappy enough about to return 404? Below is a representative SIP DEBUG trace for a call; the OPTION packet sent due to qualify=yes has the same response. -Johnathan Reliably Transmitting (no NAT) to 192.168.1.30:5060: INVITE sip:192.168.1.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4b487795;rport From: asterisk sip:[EMAIL PROTECTED];tag=as24a55bd8 To: sip:192.168.1.30 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 11 Feb 2006 00:25:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 210 v=0 o=root 703 703 IN IP4 192.168.1.2 s=session c=IN IP4 192.168.1.2 t=0 0 m=audio 19942 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called jcorgan-desk diamond*CLI -- SIP read from 192.168.1.30:5060: SIP/2.0 404 Not Found To: sip:192.168.1.30;tag=cb1ee3725d25570ei0 From: asterisk sip:[EMAIL PROTECTED];tag=as24a55bd8 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK4b487795 Server: Sipura/SPA841-3.1.1(a) Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limiting maximum runtime of echo test
Bastian Schern wrote: is it possible to limit the maximum runtime of the command echo? Use the AbsoluteTimeout application in your dialplan preceding the Echo application. http://voip-info.org/tiki-index.php?page=Asterisk%20AbsoluteTimeout -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice delivers CID even when restricted?
I can call my Broadvoice DID from a outbound caller-id blocked phone, and BV happily delivers the CID to Asterisk (and then on to my IP phone display.) I've tested with the *67 prefix from a PSTN phone to make sure it was supposed to be blocked. The number is always correct, but sometimes the the caller ID name is set to something funky (like a CO or switch center name.) I *think* this started happening after they came up from the meltdown a couple weeks ago. Is caller ID blocking implemented by sending the cid information anyway, but with a bit that says don't give to end user? I guess BV would be ignoring this bit. Anyone else experience this with BV and Asterisk? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] RE: play gsm files in windows
If I run sox (on Linux), just specifying the input and output files by with the right extensions, it will convert a raw gsm file to a wav format file while retaining the gsm compression: sox vm-youhave.gsm vm-youhave.wav This is without any additional options. The output file is playable on my Windows Media Player in XP. However, the wave file is still compressed with gsm as indicated in the Summary tab of the Properties for the output file. It shows a bitrate of 13 Kbps, and sample rate of 8 Kbps, and a format of GSM 6.10. The original file was 1320 bytes, and the produced wave file is only 1360 bytes. I thought perhaps sox was just adding the right wave file header/wrapper around the original gsm data, but a file compare doesn't bear this out. All the bytes in the wave file are new. I'm not a sound file guru--can someone explain what is happening here? Is sox uncompressing and recompressing the audio during the conversion, resulting in loss? Or, does the wave format for gsm compressed data store things differently, but uses the same underlying GSM bits, hence the differences in the file compare? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Affecting overhead with Runlevel?
Robert Goodyear wrote: So: knowing that the X11 window GUI is a resource hog, is it appropriate to use the GUI to install and configure various components, then set RUNLEVEL to 3 once all is nicely set up and running cleanly? Would this give the same effect as doing a minimal install or is the mere presence of the installed (yet not inited?) packages too heavy? That would work fine. You could still log in to the console and run the Asterisk console (asterisk -R) to watch things work, which is instructive. Corollary: if Asterisk is running as ROOT, is there any benefit to booting at RUNLEVEL 1 to prune the overhead down even further? Or is that really only for debugging or administrative issues? Yes, runlevel 1 (or single user mode as it is often called) is primarily for debugging or administration, and is set to run as few background programs (daemons) as possible. Asterisk may depend upon some of these daemons (email, cron, syslog, to think of of a few), so runlevel 3 is still your best shot. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Affecting overhead with Runlevel?
Robert Goodyear wrote: Noted. To clarify, will dropping back to runlevel 3 still ensure a smaller set of processes that would be as non-intrusive as if I had installed Linux with console/command line support only or would there still be stuff hanging around that's inextricably there because I had _at one time_ installed and run the GUI? No, runlevel 3 typically doesn't include any graphical console processes. Having them installed but not running only wastes disk space; there would be no difference to Asterisk from a CLI only installation. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outbound dialing issue with FXO
John Novack wrote: ...along with detection of stutter dial tone on analog lines. This would be somewhat useful. When there is stutter for whatever reason, outbound dialing must be delayed a couple seconds. Some times my PSTN line would roll-over to the telco voicemail before * would answer, someone would leave a message, then outbound dialing would fail until the PSTN voicemail was cleared. (Now my * answers on first ring so this isn't a problem anymore.) Would this have to be done in the Zaptel hardware driver, the chan_zap code, or somewhere in the mainline code? I've seen the tone detection routines in dsp.c, could these be brought to bear? (Answer in -dev if this gets off -user subject too much.) -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Newbie on IVR
Mike-Olumide, Johnson wrote: I get fascinated when I dial someone and get an IVR play for accounts department press 1, for sales, press 2 or hold the line for an operator and then have MOH play before the line is picked up at the desired extesion. You'll find a simple example of how to accomplish this at the voip-info.org website: http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu This website has a large amount of information for using Asterisk, follow some of the See Also links at the bottom to learn more. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outbound dialing issue with FXO
Mike Clark wrote: However, outbound calls are hit or miss. Sometimes they work fine and other times we get a you must first dial a 1 or 0 message back from telco when dialing out standard POTS lines. Did you get this working yet? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Outbound dialing issue with FXO
Mike Clark wrote: We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a you must first dial a 1 or 0 message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six Polycom phones are on the system. Any ideas on this one? Just a shot in the dark...I was getting this same thing with a TDM11B card. Turns out I needed to add a pause before dialing on the FXO port; the first digit was getting lost randomly. Adding a 'w' before the extension expression in the Dial command fixed it entirely. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Gerenal settings conufsion
Jeffrey Starin wrote: Jonathan! You don't know how much that simple explanation has helped me understand Asterisk. Well done. Well said. And to the point clearly. I would hope this could find it's way onto the Asterisk Wiki and be the *first* thing someone reads when looking at the documentation about sip. Thanks a million! blush There are many things I've found in Asterisk so far that take a while to wrap one's brain around. Once the effort is made, though, it's definitely worth it. Hopefully one day the learning curve won't be quite as steep. You'll find that once things start falling together in your mind, it gets a lot easier. And really fun, too, if you're into that sort of thing. Reminds me of early-90s Linux--and look where that is now. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Broadvoice: No Service, No Email reply but charging the credit card still works
Ronald Wiplinger wrote: I cannot email them, I cannot call them, I do not get an answer, but the credit card is still charged, although NO phone calls are possible anymore, ... Hmm. I called them twice yesterday to ask questions, the queue wait was less than a minute in both cases. First time was via their own service, second time was via my cell phone. Their techs were very friendly and accommodating (but not the most knowledgeable.) My service is working normally. In fact, now that things have settled out with their new partners, it seems to be working *better* than before--faster call completions, better voice quality. Network probes show 1% packet loss to their lax and dca proxies and back over a 48 hour period. They emailed me some follow up information after my last call; it arrived a few minutes later. So yes, they are still in business. (They were within moments of losing *my* business when all the lights came back on last week. Guess I'm just a sucker for new technology and unlimited free phone calls :-) Can you elaborate on what is happening with you? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Home Usage
Kerry Garrison wrote: PSTN line 1 - Parent's line PSTN line 2 - Kids' line PSTN line 3 - Bussiness line VOIP Provider 1 - Broadvoice VOIP Provider 2 - VOIPJet A call into Line 1 answers For Kerry press 1, For Karen press 2 A call into Line 2 answers For Taylor press 1, For Chris press 2 A call into Line 3 answers Thank you for calling tech data pros blah blah blah All outbound calls got out broadvoice first, voipjet as a backup, and the pstn lines as a third backup. Is that trick enough? Add soft phones for your wife's relatives scattered all over the world with PCs and Internet access. Give them voicemail, and outbound access to your free stuff. Achieve marital bliss. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP Gerenal settings conufsion
Jeffrey Starin wrote: I have a little confusion about the general settings (other than the register values) in the SIP General area. [snip] However, I'm confused as to the purpose of the general settings -- to what or which connection do they apply? Since the context suggested for the general settings is something like nothing to avoid unwanted sip calls, I'm confused as to the purposse of values in the general section since all parameters for communications with VOIP providers is contained in the contexts such as I specified above, i.e., [FWD] or for example [BROADVOICE]. Can someone shed some light on that for me? Well, your confusion is understandable, given the way sip.conf works. Parameters which affect incoming calls are not separated from those that affect outgoing calls, so it's easy to get mixed up (well, for me, anyway.) In the [general] section the parameters become the defaults used unless overridden in a specific peer section. Also, if an incoming or outgoing SIP call doesn't match a specific peer section, these parameters get used. So, for example, if you don't want any incoming SIP calls that aren't from a known provider, you can set the default context to something innocuous as you describe above and the call will get rejected as a non-existent context. This is what you described in our original mail. But also in this [general] section are settings for *outbound* calls using SIP that aren't using a specific peer section. This can be done with the Dial command, using a dial string such as IP/[EMAIL PROTECTED] where provider.com is not listed in sip.conf. This might happen, say, in the case of using the ENUM lookup capability, where the outbound SIP address of a phone number is determined dynamically at call time rather than pre-configured in extensions.conf/sip.conf. Personally, I'd like to see this changed so there are two 'general' sections--one for default parameters to use unless overridden when there *is* a peer section below, and a different one to describe parameters to use when the remote peer is not previously known. I know there are ways to accomplish this with the existing sip.conf structure but it seems very counter-intuitive. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ITSPs with good phone support
With the recent service outage at Broadvoice, there has been a lot of discussion here, on broadband reports, Voxilla, etc., regarding whether VOIP is mature, or ready for the masses, etc. One particular point I've seen repeated, and with which I agree: we're willing to deal with less than five 9s, even one or 2 9s, as long as we have good communication regarding the issue and its resolution. In other words, good customer relations are as important or even more important than the highest quality of service. This is no great revelation, but I don't see it practiced very much in this industry. We read many comments about long hold times, dropped calls after waiting in the queue, tech support agents who misdirect or hide internal issues, etc. Personally, I could almost live with the bizarre happenings at Broadvoice if there was an official channel of communication outlining what was happening and estimates on return to service. While I recently on this list have been a supporter of them (growing pains, not incompetency), recent events, with zero communication, have worn me down. My money is going elsewhere. As a counter example, I recently switched to Sonic.net DSL service. They turned the service on almost a week before they said they would, and I called the support line to get some information that would normally arrive by mail but hadn't yet. There was no queue! After three rings, it answered, and the person who picked up was able to answer all my questions, was not hurried, and could even answer my Linux and Linksys questions, even though this wasn't supported. I think they must pay these people a lot more money than industry standard, as their customer skills were outstanding. On a $45/mo. service, no less. I called back when something wasn't working right, and got the same experience, with the problem fixed while I waited on the phone. This has nothing to do with the maturity of the technology or newness of the industry, this was just a good investment in a customer service department of the business. Where do the rest of you weigh in on this? Are there any VOIP to PSTN gateway companies with this same sense of focus? VOIP *isn't* mature and probably *isn't* ready for the masses, but we early adopters seem to be able to live with these things if there is appropriate communication going on. Who's the leading contender for customer service of the year award among the dozens of providers that show up on the Wiki? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Who's happy with their voip service?
JD wrote: Inbound calling has been down for 2 days. Just FYI, mine is back up (408-903) as of about five hours ago. I did just speak with a (Broadvoice) support tech on an entirely unrelated matter (40 min. hold time!), mentioned mine was working, and he seemed to think things were coming back in stages. I've had them for two months now. People may recall a series of emails regarding packet loss through their PNAP link to Sprintlink (my ex-ISP backbone.) I ditched the Sprint BBD fixed-wireless service, got Sonic.net DSL, and have been enjoying pretty high quality voice service since. The packet loss rates at PNAP still show but I think now this shows it's an artificial measure (intentionally dropped non-VOIP packets, all the other potential reasons hashed about in that thread.) In spite of the service outages and long hold times for support, I still want to give them the benefit of the doubt (and my $25 monthly.) It still seems like growing pains vs. incompetency. I tried their web interface to change DIDs, as they now have them in my home area code. The effect was instant, I reconfigured sip.conf with the new number and secret they provide, and something like 3 minutes later was using the new DID. So some things do work well. Wish they did IAX. And ILBC. Not that important to me right now, though. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Light weight and slimmed Asterisk
Kumara Jayaweera wrote: Sorry for the numerous postings. but How could I slim my Asterisk PBX. Really I don't need such modules like Ex. chan_modem.so. Becouse, I don't have any special hardware. Please, could I hope your various suggetions in this regards. brief me your idea. See the Wiki under Asterisk Slimming: http://www.voip-info.org/tiki-index.php?page=Asterisk+Slimming The basic idea is to either: 1) Have 'autoload=yes' in your modules.conf, then use 'noload' to indicate modules not to load, or 2) Have 'autoload=no' in your modules.conf, then use 'load' to indicate which modules you need. The Wiki page has an example of each. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Multiple Servers and 1 Central Voicemail
Josiah Bryan wrote: Why not just NFS mount the /var/spool/asterisk/voicemail directory from a central server? That way, all servers share the same spool and the MWI will get reflected on all servers. Does * use any form of locking to maintain the integrity of the sequence number for voicemails in a given mailbox? What happens if two different servers want to record a voicemail in the same mailbox at the same time, would they both grab the same next highest number to use? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Multiple Servers and 1 Central Voicemail
Jason Brown wrote: Trust me though, I promise, 1 central VM store does work and work well in an asterisk environment. But I haven't seen addressed the issue of two or more servers sharing this central VM store, when running the Voicemail application. Sure, MWI should have no problems in the scenario you described, but what about two or more Voicemail applications running on different servers, sharing the /mnt/asterisk/vm tree, and simultaneously adding and removing voicemails from a users mailbox? What if a user is checking/deleting voicemail on one * server while another person is dialed in a different server, leaving him a voicemail? Does * handle this safely? There's all sorts of issues of contention I can think of, and all of them have well-known solutions based on file locking. But does * use them? I'm not trying to over-complicate things. Someone proposed sharing a voicemail filesystem hierarchy between multiple * servers and I'm wondering if * has been programmed to safely handle the concurrency problems that would occur. Guess I'll go digging through the source code. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] BroadVoice Inbound Not working ..
Brian Watters wrote: Ok .. It appears we have outbound calls working well via our SIP trunk and BroadVoice .. However inbound just plain does not work, What happens is when you call our BroadVoice number it tells you the person you are calling is not available .. Never makes it to our Asterisk? .. Any ideas where to look and or what to tweak? If the incoming SIP connection really isn't making it to your Asterisk (verify with 'sip debug' command on console), it is likely because BV doesn't know where to connect to. * accomplishes this by a process called registration. In your sip.conf, near the top in the [general] section, you should have a line that starts out: register = If this is there, can you send it (and change the password to something innocuous?) If this line is there, you can see whether it succeeded by typing 'sip show registry' at the console. If that shows the state of registered, and your sip debug output shows an in coming connection request from BV, then there are further steps for debugging, but start with the above. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] broadvoice config problem.
Craig Simon wrote: Looking for 100 in from-broadvoice It looks like * is searching for extension 100 in the 'from-broadvoice' context, not finding it, and sending a 404 back. First, you can create a extension 100 in that context in your dialplan, then see if that allows the call to come through. Second, why would * be looking for extension 100? Can you post your sip.conf? Do you have a /100 or something on the tail end of your register statement? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] broadvoice config problem.
Craig Simon wrote:
The 100 is an extension I created for my softphone to log into.
* is tricky with terminology.
You didn't create an extension 100, you created a SIP peer/user named
100, which the softphone connects as.
Extensions (that are within contexts) are lists of commands that *
will execute if that extension is dialed. So what you want is to create
an extension 100 that will have the list of commands to dial your SIP
phone. Your current set up is trying (by virtue of the Dial command in
the from-broadvoice context) to find extension 100 in the default
context, which is where you would have the commands to dial your SIP phone.
This is an unnecessarily convoluted way of doing things.
Here's an alternative:
[from-broadvoice]
exten = 100,1,Answer
exten = 100,2,Wait(1)
exten = 100,3,Dial(SIP/100,25)
exten = 100,4,Voicemail(u${EXTEN})
exten = 100,5,Hangup
exten = 100,104,Voicemail(b${EXTEN})
exten = 100,105,Hangup
Here the sequence:
1) Incoming SIP call from Broadvoice lands you in the from-broadvoice
context, looking for extension 100. (Why this is 100 is still a mystery,
but it appears to be working that way, so we'll assume that's correct
and move on.)
2) * starts executing the commands in 'priority' order, meaning it will
answer the Broadvoice call, wait a second, then attempt to dial SIP peer
'100', which is your soft phone.
3) If you answer the phone, * bridges the two channels and it's done.
4) If you don't answer the phone or for some reason it can't connect to
your soft phone, it will instead connect the Broadvoice call to the
voicemail for mailbox 100, and play the unavailable message. When the
voicemail application ends, it hangs up the two channels and is done.
5) If your softphone indicates busy, * will jump to n+101 (n being 3
here), which means it will connect the Broadvoice call to voicemail for
mailbox 100, but play the busy message instead, and then hangup when the
voicemail application is done.
Later, you can change things so that you have and IVR system instead,
and start to use macros, etc., and then you'll learn about the Goto
command and all. But first get this simple thing working.
You didn't include in your email the register = command, can you show?
-Johnathan
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Re: [Asterisk-Users] broadvoice config problem.
Craig Simon wrote: Thank you, thank you, thank you! That was it! Thanks alot for the description, it made the call flow much easier to understand. You're welcome. I only discovered Asterisk about a month ago myself, and understand first hand how difficult it can be for the uninitiated. Asterisk is a lot like Linux was in it's early days--extremely powerful, lots of fun to play with, frustratingly fragile at times, haphazardly documented, supported by a community of experienced people that can sometimes be hard to communicate with, continuously improved by a motivated herd of sharp developers, and, I think, destined to make as much a difference in the telecoms world as Linux has already made in the OS world. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] broadvoice
Dalon Westergreen wrote: BV allows unlimited incoming, and up to 3 outgoing. My understanding is that they intend to charge for more 3 outgoing, but have not done so at this time. This is good to hear--do you have anything from BV that documents this? Also, being relatively new to *, I don't know if there is an obvious way to enforce this in the dial plan, returning congestion if three outgoing calls to BV were already taking place. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoIP Provider problems
Bob Goddard wrote: The apparent packet loss you are seeing may be just fine tuning of the routers in question. This is the conclusion I came to as well; however, with the way PingPlotter works the router is not sending ICMP unreachables but rather ICMP TTL expired responses. In any case, the routers in question may either be: 1) ...intentionally discarding the received UDP ping packets (these are not ICMP pings, but rather UDP packets with TTL down to zero when they get to the router), because the router has better things to do. 2) ...throttling the ICMP TTL expired responses to a certain rate per period of time, as you suggest. This would appear as packet loss. 3) ...actually congested, with the received UDP pings (and other types of packets) getting discarded on the input side at the rate shown in the data. I wish there was a way to measure 3) without being affected by 1) and 2). I agree then, that PingPlotter is not a highly accurate way to measure path quality. Still, though, looking over the data for a couple days now it is easy to see cyclical patterns that go from 1% to 30% (PingPlotter measured) loss, and an easily seen correlation with the voice quality of my outbound Broadvoice calls. Interestingly enough, switching from a Firefly soft phone on my workstation, using IAX2/ulaw, to an analog phone-TDM400 FXS port right at the Asterisk server has made a big difference. So some of the perceived crappiness was in the soft phone-Asterisk path and was probably being exacerbated by the network loss on the net or at Broadvoice's router. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoIP Provider problems
Rich Adamson wrote: In other words, as the ttl value is increased and additional icmps are sent, you might see what you believe is congestion, but you still don't have any clue as to whether hop #2, #5, or #10 actually was involved with that congestion. Sure. But there is a way around this. The traceroute-style statistics gathering technique that PingPlotter uses tries all the hops at the same time and plots the return rate for each one. So a 10 hop path has 10 packets go out, with individual packet's TTL set to expire at each hop. Done over and over again and averaged over many probes, you get a very clear picture. Packet loss at one node affects all the probes to that node and further ones, resulting in an increasing loss rate as you go down the path. For example: Hop Loss 1 0% 2 1% 3 1% 4 5% 5 5% 6 6% 7 15% 8 15% 9 16% 10 16% It's easy to see there is a big problem between hops 6 and 7 and a smaller problem between hops 3 and 4. With the broadvoice router I was seeing (at first) a jump from 0% to 9% at my local ISP, then small increments over the next 10 hops until it was at about 14%, then a big jump to 29% at the last hop. The data has varied cyclically between as high as the above and as low as 1% all the way across. Right this very moment, it is 2% within my ISP, still 2% all the way to PNAP, then a jump to 14% at the broadvoice ingress router at PNAP. Again, temper the above with the fact that the packet loss may be intentional, and these statistics not representative of real RTP traffic, as per my previous message. But I can predict with high accuracy what the caller on the other end of my broadvoice call will say about my voice quality based on the last number I see for the broadvoice ingress router. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoIP Provider problems
Johnathan Corgan wrote: First off, I have Sprint Broadband Direct internet service, a fixed wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink. So I know I'm in for trouble anyway. The broadvoice edge router (63.251.209.126, their lax site) is another 11 hops away. One hop before that, the packet loss rate has gone up to 13%, so the Internet adds another 4% to my sucky ISP connection. Round trip time to this point is 200ms, so-so but livable. Here's the kicker: Reported packet loss from broadvoice, one additional hop, is a whopping 29%. So between the last Internet router (bbnet2.lax.pnap.net) and broadvoice's edge router, there is an additional 16% loss. Just an update after about 12 hours of data--the data above was worst case. During off-peak hours in the middle of the night the packet loss at my ISP was effectively zero, and only 3% along the way to broadvoice, with a 75ms round-trip time. Broadvoice edge-router still reports 28% packet loss though, and an additional 30ms RTT increase for this last hop. So I even more strongly suspect (or just really hope) they are preferentially discarding non-RTP traffic in favor of voice traffic. I did discover that the multi-second outages are at my local ISP, not at Broadvoice--for some reason Sprint BBD can take up to 4 seconds to respond to a ping, so something is really wrong there--but is there a way to do this type of testing in a more rigorous and controlled fashion? -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoIP Provider problems
Joseph Gutowski wrote: I installed PingPlotter, switched to UDP just to be the same as you, and ran it against sip.broadvoice.com. Absolutley no problems, no packet loss at all. Well, that's good to hear. I then used the 63.251.209.126 that you posted, and it was awful (at least it appears awful). I have reliable 20% packet loss at each of two Verio hops (nothing lost at the far end). Okay, happy to see independent confirmation of this. I did traceroutes on all of the Broadvoice proxies, and I didn't get pushed through PNAP. I wonder why your packets seem to reliably following that path when it's so bad. I mean the whole point of routing through PNAP is to increase quality, no? And from my understanding they're supposed to have a magic fuzzy logic to dynamically reroute around problems. Your results suggest a more widespread problem than one customer can't have nice VoIP calls -- you'd think Sprint wouldn't be routing through PNAP. Well, you're right, Sprint is going through sprintlink.net - PNAP - BV, no route changes during the day since I started. Not a lot I can do about that, unfortunately. And I also hope your VoIP connection is wired if you're getting 9-10% loss on the wireless before you even leave the LAN. If you're starting off with a loss, it's just going to make the natural losses on the net have an even worse effect. It appears I happened to pick the most congested time to measure, and got 8-9% packet loss on my Sprint uplink. That's the wireless, as in a fixed wireless MMDS rooftop dish link to a mountain top about 15 miles away. It turns out that off peak there is zero loss over this link and typically it is only about 2-3% loss. So it's not as bad as it first seemed. On the premises it is all wired and first router is always zero packet loss. As I write this the trailing 10 minutes of data shows an aggregate 9% loss to BV with 3% of that on the Sprint BBD uplink side. This is much better than my first tests, and my SIP calls through broadvoice show the difference too. Anyway, I haven't tried the other broadvoice proxies yet, I'm really hoping at least one doesn't have PNAP on its path. (At least I can be thankful I haven't run into any of the weird NAT or authentication issues that have been discussed--worked great first time.) At the time I got this wireless link (which with a 4Mbps downlink, is pretty sweet for typical traffic patterns), there was no DSL in my area. Now SBC has service, but I've yet to really look into it. (As an aside, got my TDM400 card today, installed, and have the FXS port working with an analog phone. Woohoo. FXO next!) -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] VoIP Provider problems
Robert Terzi wrote: The best tool I've found for monitoring connections, routes, congestion, is called PingPlotter. http://pingplotter.com/ It's a shareware visual traceroute. It continually graphs the traceroute style responses. There is a scrollable timeline to view how things change. You can get raw data out of it as well. It records changes in routes. Thanks for the excellent link. I've had Asterisk on a home network and Broadvoice for a couple weeks now. IAX2 calls between Firefly soft-phone on my desk and other soft phones directly on the net have worked fairly well, but reported voice quality when going out over broadvoice to the PSTN has really stunk, making it only marginally useful. So I've downloaded this utility and am now tracing out sip.broadvoice.com, using UDP (as my ISP filters icmp.) Actually, the trace doesn't get past broadvoice's edge router, so I replaced the final IP address with that of the edge router itself so I could see the data instead of destination unreachable. Anyway, with only 20 minutes I've data I'm seeing some rather disappointing results. First off, I have Sprint Broadband Direct internet service, a fixed wireless setup with a 2-5 Mbps downlink and a terrible 128 kbps uplink. So I know I'm in for trouble anyway. First hop off my home lan over the wireless starts at about 9% packet loss. Sucks but for normal TCP based stuff (email, web, ssh, etc.) life goes on but just a little slower. The broadvoice edge router (63.251.209.126, their lax site) is another 11 hops away. One hop before that, the packet loss rate has gone up to 13%, so the Internet adds another 4% to my sucky ISP connection. Round trip time to this point is 200ms, so-so but livable. Here's the kicker: Reported packet loss from broadvoice, one additional hop, is a whopping 29%. So between the last Internet router (bbnet2.lax.pnap.net) and broadvoice's edge router, there is an additional 16% loss. No wonder my outgoing voice to the PSTN is choppy, filled with several second gaps, and makes people laugh at me for spending $20 a month on VOIP. I admit I can help things a bit by getting an ADSL or SDSL link with a better provisioned uplink, but even if I had 0% loss to broadvoice, their own net connection seems seriously under-provisioned. One thing might be affecting this and make these numbers suspect--broadvoice might have QoS on the edge router such that non-RTP packets get lower-class status, so my UDP pings are artificially dropped in favor of real RTP traffic (actually, I'd be doing this if I were them.) Anyone care to comment on how realistic a test this is? I'll do these tests for a few hours and hit the different broadvoice proxy networks and see if there is a difference, and compare to loss rates for other sites over my ISP uplink. Anyway, my Digium 11b card comes in tomorrow, so I'll be off to more fun setting up the IVR and voicemail, etc., for my home line off the PSTN...love this Asterisk thing. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
