Re: [asterisk-users] names of SIP aware firewalls
Sonicwall, but I have no idea if it really works. Jerry Jones wrote: Intertex Not cheap, licensed per number of users But seem to work great and have some nifty tools very confusing picking models though On Nov 5, 2006, at 3:54 PM, Erick Perez wrote: Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? ___ --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] FXO Cards vs. Channel bank with T1
You'll find the cost of a PRI varies dramatically from one telco to another. I've heard numbers in one case where three analog pstn lines cost the same as a PRI, another case where 16 analog pstn lines cost the same as a PRI. And, having worked in the telecomm industry for many years, there are still a very large number of telco's that do not support PRI's at all. Rich Dovid B wrote: Looking at the number's now it seems that a T1 will be more. Anyone here sell PRI's ? - Original Message - From: Jay R. Ashworth [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 31, 2006 9:38 PM Subject: Re: [asterisk-users] FXO Cards vs. Channel bank with T1 On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? If you need enough ports to make a T-1 card cost-efficient, then you might oughtta be looking at an Ethernet to FXO media gateway instead -- assuming you need analog interfaces. FXO side, why not just go T-1 or PRI? Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think '87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stopping putgoing calls after working hours
I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk?? You need to better describe your objectives. If you really mean stop all calls (including emergency calls), that's easy. If you mean stop all calls that cleaning folks initiate (usually not employees), that just requires some extensions.conf changes to force the user to enter an access code before a call can be placed. (Just don't advertise that access code anyone that you don't want making calls. If your talking about a fairly major security issue (such as your users call forwarding their phones to the brother-in-law after normal hours, you'll probably need to disable call forwarding on the phone itself. If your talking about primarily managing expenses, use the CDR detail to generate a personalized report for each employee show this calls make between 5pm and 7am, and forward that report to each employee (and cc: the manager). That's usually enough to significantly cut those calls. If you don't have a policy relative to use of company assets (phones PC's) for personal use, you might put one together and reference that policy in the morning CDR detail report. (I'm sure at lease some of those calls are likely legitimate calls, so cutting all calls is not likely a workable solution. ___ --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] DID failover
Todd- Asterisk wrote: I'm setting up an asterisk server where an administrator will not always be available in case of problems. While I expect problems to be rare, I need to be prepared. We're thinking of VoIP DID's and SIP phones so it's an all TCP/IP network. We could get a second server to substitute - What is involved in 'transferring' or 're-registering' the DID incoming lines to a second server in case the primary is down? If there a better fall-over method? I'm looking for the easiest way for the un-educated sys-admin-apprentice to handle it. The system doesn't exist yet so any suggestions are appreciated. I recognize I'll need to modify the SIP phones- I'll figure that out later. thanks in advance One of the simplest ways to accomplish this is to use an APC power strip with SNMP control. (Each of the power outlets on the power strip can be turned on / off remotely via an snmp command. With this rough approach, stop the 'broken' asterisk server and start the backup server (via the power strip control), and wait for the system to come up. If both asterisk systems are configured absolutely the same (eg, same *.conf entries, ip addresses), then when the system comes up, it will 'register' with your sip or iax provider. The sip phones will likely take a little bit longer to come up due to arp cache timout values within the sip phones. I've not tested any of the sip phones to see what the default timeout values have to be, but it will vary by manufacturer. (Microsoft PC stuff is generally around two minutes.) As soon as that cache value timeouts out, the sip phone will register (with the new server) and should be totally functional. If at some future time you need a T1 or PRI on the system, someone manufacturers a T1 relay that will swap the T1 from one system to another. The downside to this approach is that you have to wait on each device's arp cache timeout value (including routers, dsl moems, sip phones, ATA boxes, and any other device that is required in you fully working system. Very few of the voip devices allow you to set the arp timeout value. In very general terms from a historical perspective, abruptly shutting down power to a linux/unix box is not is not an acceptable practice. However, the newer systems are far more tolerant, and for emergency purposes, its probably not that bad as the last step. If you read over some of the archives, there are other ways that involve redundant servers, heartbeats, load sharing, reserving a valid extension number that would kick of scripts (etc) to swap boxes. Each have their advantages, disadvantages, and costs. ___ --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] SPA942 quality for a Bank
Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has been purposefully configured with only basic telephony functions. Oh... someone mentioned the headset (no handset) pin jack is only for the microphone (and not the speaker) which would seem very odd. Anyone using a headset with the 942 where both the microphone and earpiece function fully? Any thoughts? ___ --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] SPA942 quality for a Bank
Tom wrote: At 02:30 PM 10/15/2006, you wrote: Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. We have been using Cisco hard phones with Asterisk for over two years. Our longest experience has been with 7960g phones although recently I switched to the 7970g for evaluation. Several of us have tried the SPA942's for about one month. Personally I was real happy to go back to a Cisco IP phone. The sound quality is useable but just not as good on the SPA942. And the speaker phone on the SPA942 is poor enough quality that it is barely usable. Yes, we see about the same here comparing the old BT102, Cisco 79x), and the spa942. Given most of the demo's thus far are with small banks, price weights very heavily in their mind. We include the SPA-942 in our side by side demos for prospective customers. So far they are buying Cisco and willing to pay the higher price. The sales pitch tries to address the sip licensing costs on the Cisco 79x0's, and when that's added to the base refurb cost, the banks seem to move quickly to the 942's. We only have four of the SPA-942's and have not seen any failures in our limited use. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has been purposefully configured with only basic telephony functions. Oh... someone mentioned the headset (no handset) pin jack is only for the microphone (and not the speaker) which would seem very odd. Anyone using a headset with the 942 where both the microphone and earpiece function fully? We have used them with VXI headsets and the microphone works fine. Good. Who mentioned the 'mic only' must not have had the correct headset/plug for the spa942. ___ --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] FYI - Polycom SoundPoint IP 301 Denial of Service]
FYI. TITLE: Polycom SoundPoint IP 301 Denial of Service SECUNIA ADVISORY ID: SA22266 VERIFY ADVISORY: http://secunia.com/advisories/22266/ CRITICAL: Less critical IMPACT: DoS WHERE: From local network OPERATING SYSTEM: Polycom SoundPoint IP 301 http://secunia.com/product/12229/ DESCRIPTION: A vulnerability has been reported in the Polycom SoundPoint IP 301 VoIP Desktop Phone, which can be exploited by malicious people to cause a DoS (Denial of Service). Sending a long URL to the embedded HTTP server or using the Nessus http_fingerprinting_hmap.nasl script can cause the phone to reboot. Additional, it has been reported that the TCP port 42 is open and accepting connections. The vulnerabilities have been reported in firmware version 1.4.1.0040. Other versions may also be affected. SOLUTION: Reportedly, this does not affect the firmware version 2.0.1. PROVIDED AND/OR DISCOVERED BY: Shawn Merdinger -- About: This Advisory was delivered by Secunia as a free service to help everybody keeping their systems up to date against the latest vulnerabilities. Subscribe: http://secunia.com/secunia_security_advisories/ ___ --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] VOIP with PSTN backup
Brian Candler wrote: I'm looking for a way to set up a VOIP network in branch offices where one or more phones have lifeline capability, i.e. can place calls if the IP network or VOIP service dies, or even if power goes down. (I'm thinking of business continuity here, not just emergency services) This seems to limit my choice of products somewhat, and I was wondering if anyone had recommendations for use in this scenario. The approaches I'm thinking of are: (1) Use an ATA with PSTN passthrough or FXO port, and connect an old analogue telephone to the FXS port. In this case, the analogue phone has lifeline. If there's a true FXO port then PSTN calls can in principle be routed to/from other VOIP phones in the office (but see below) There seem to be a few to choose from, although far fewer with a true FXO port. (2) Find a VOIP phone with integrated PSTN or FXO port In this case, the only one I have found so far by searching the web is Clipcomm CP101. I have also read that many FXO devices tend to be badly implemented; in particular, on seeing ringing voltage, they actually pick up and answer the call, instead of sending off a SIP INVITE and waiting for the OK before connecting. I'd certainly like the device to behave properly in this regard. As a second part of this question, it would be extremely desirable if the backup PSTN service were available to all the phones in the office. That means: (a) incoming PSTN calls could ring *all* the VOIP phones in the office, not just the one phone or ATA connected to the PSTN line; and (b) any VOIP phone could route a call out over the LAN to the local FXO PSTN port, e.g. by dialling a prefix to access it. This isn't so essential but it's definitely desirable. Any recommendations for how to do this too? A large number of offices is going to be involved, and I want to keep as much switching intelligence centralised as possible, both for ease of management and to keep the cost down. That is, I don't want to install a PC + TMD400P + Asterisk in each location, but just a small media gateway or VOIP phone. However I can see that the incoming ringing issue will require call forking, so I am happy to install an OpenWrt box running Asterisk or siproxd or whatever in each site. Being diskless and low power should mean little maintenance is required. But such a box isn't going to be able to take an FXS/FXO card, so I'll still rely on an ATA or VOIP phone to present a PSTN interface. So that's the key part I'm looking for. Finally, the devices must be robust (i.e. not need power cycling every 24 hours) and centrally manageable. I think that's about it - many thanks for your ideas and experience! If you get real serious about this, then do a risk assessment for each component involved in the end-to-end communications system. The risk assessment should include an analysis of each component answering questions like: 1. What's a reasonable business down time for the communications system (and that answer is not zero) 2. How important is the component (high, medium, low) 3. What's the likely restoration time for the component 4. What are some of the potential causes for a component failure etc, etc. Once that is done, I think you'll find that you can prioritize which assets need to be addressed in what order. For example, a fiber seeking backhoe will likely disable all forms of communications (eg, analog and digital). Therefore, trying to locate a phone (or ATA) with an analog fxo port is of no value. Finding an alternative carrier maybe based on some form of wireless service, cable broadband, etc, might be a reasonable approach. Some companies will actually bury telecomm communications facilities into a building, arriving from two distinct locations, thus reducing the exposure to the fiber seeking backhoe. ___ --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] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute
Noah Miller wrote:
You can also change some settings in the zapta and zaptel
config.. to reduce
echo and interference on the line..
This is the most important thing here - what does your zapata.conf look
like?
zapta.comf
switchtype=national
This is not necessary in your case. It pertains to PRI lines, and not
the POTS lines you have.
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
You may want to turn each of these off, in turn, for testing,
especially the echocancewhenbridged.
You can also tune the echocancel setting in terms of taps (a tap is
one sample from the data stream per second). You can use the values:
16, 32, 64, 128, or 256 ('yes' just means 128).
Might also try echotraining=800. That parameter causes the zaptel code
to wait 800 milliseconds before pulsing the pstn line, and that pulse
return is used to preload the software echo canceller to some reasonable
starting point. Not usre if this will have any impact on your problem,
but might be worth a try.
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Re: [asterisk-users] Outbound FXO call, getting You must first dial...
Nick Ellson wrote:
I am not sure what I might be set up wrong, but dialing out with my
Zap/1 port seems to alwyas get the You must first dial a 1 when calling
this number message from what sounds like the actual PSTN. My
zapatel.conf and extensions.conf bits below. Any advice? (I do receive
inbound calls, and it does sound like I am getting the PSTN error. I do
notice that when I get an inbound call, I have 5 secs of sevear static
before it suddenly becomes clear.. could that be happening on the
outboud as well munging the first few digits?)
signalling=fxs_ks
language=us
context=inbound_qwest
sendcalleridafter=2
callerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
channel=1
exten = _9.,1,Dial(Zap/1/${EXTEN:1},60)
You should probably do a little research before posting questions like
this as its been answered many many time.
The problem is that some pstn central offices are not ready to receive
dtmf digits as quickly as what asterisk sends them. So, an option w
has been added to the Dial command to instruct asterisk to wait about
200 milliseconds before sending dtmf. Try something like this:
exten = _9.,1,Dial(Zap/1/w${EXTEN:1},60)
and notice that lower-case w in the string. If that doesn't fix the
problem, try two ww's in a row.
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Re: [asterisk-users] No Dialtone
If you've messed up in connecting telephone lines to the wrong module, the ringing voltage sent to a fxs module will destroy it. You would need to replace the module. Eddie Johnson Jr wrote: Yes, I have and I received the following: In zapata.conf your first two channels should be fxs_ks because the first two modules are FXO mdoules. Your last two channels should be fxo_ks because the second two modules are FXS modules. For the TDM400P(TDM 22) the FXS modules work with the phone. The 3 port is for the line. So I unplugged it from port 3, and plugged the analog phone in port 1, made the changes to the channels and set immediate=no, restart the server and activated asterisk. Nothing, my friend. Any more suggestions, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Francesconi Sent: Friday, October 06, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No Dialtone Did you set immediate=no in zapata.conf? Francesco Eddie Johnson Jr wrote: Hello, I have the following setup: 1. Ubuntu Dapper Server 6.06 plus latest patches 2. Asterisk 1.2.11 3. libpri 1.2.3 4. Zaptel 1.2.8 5. Digium TDM22 (TDM400P) 6. Analog phone plugged in port 3 7. The wctdm, zaptel modules load at startup, I type asterisk as root and it is activated. 8. I check the Channel Map and I have the following: Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. I can ssh into the server and remotely connect to the server. Great! The card is not connected to an outside line as of yet but I have no dialtone on the phone. I spoke with a rep. at digium and was told a dialtone should be there. Zaptel.conf : loadzone=us defaultzone=us fxoks=1,2 fxsks=3,4 Zapata.conf: ;FXS Modules signalling=fxo_ks channel = 1,2 ; ;FXO Modules signalling=fxs_ks channel = 3,4 I made sure the card is not sharing an IRQ, I checked the hard drive and all is well. I load zttool and get the following: cat /proc/zaptel/* Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 Any suggestions? Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Dialtone
Jay R. Ashworth wrote: On Sat, Oct 07, 2006 at 10:31:16AM -0500, Rich Adamson wrote: If you've messed up in connecting telephone lines to the wrong module, the ringing voltage sent to a fxs module will destroy it. You would need to replace the module. I'm going to stick my neck out here, and opine that any FXS module that would be destroyed by receiving ringing voltage is *incredibly* poorly designed, and very probably wouldn't pass Part 68. Shouldn't, certainly. Try it and see what happens, and report back. ;) ___ --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] CALEA support within asterisk?
Does anyone know if asterisk currently supports the US government's Communications Assistance for Law Enforcement Act (CALEA) regulations? If not, does anyone have this item on their To-Do list? For those that are not familiar with CALEA, it's the governement's way of intercepting or monitoring voice communications (presumably with a court order) for law enforcement personnel, etc. The broadband / ITSP compliance due date is May 14, 2007. The CALEA implementation and compliance for pstn central offices is complete (with some exceptions), and required software development efforts by each of the central office switch vendors. Rich ___ --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] CALEA support within asterisk?
Inline... On Tue, Oct 03, 2006 at 12:13:27PM -0500, Rich Adamson wrote: Does anyone know if asterisk currently supports the US government's Communications Assistance for Law Enforcement Act (CALEA) regulations? If not, does anyone have this item on their To-Do list? Why in hell *would* anyone? I don't think CALEA (to, not for :-) applies to anything smaller than a CO switch anyway, does it? Yes it does. Firstly to address the original poster: http://hraunfoss.fcc.gov/edocs_public/attachmatch/DOC-260434A1.doc CALEA can in this case be implemented by your provider alleviating you somewhat. Depending on how you infer legalese. As the OP of this thread, I'm involved with an itsp operation that includes asterisk with links to regional/national itsp facilities, PRI's to local pstn facilities, and broadband sip/iax connections to residence and business customers. I don't think the legalese will be justification for not providing calea support. CALEA outside of sniffing, facilitates recording information (CDR's etc.) , so setting up a designated machine (syslog perhpas) and saving the logging information(/var/log/asterisk/*) from Asterisk will likely suffice. I've not dug into the calea requirements indepth as yet, however I believe it does require real-time call monitoring (eg, audio), cdr-like records, and some form of reporting (unknown what reporting truly means in this case). So, my initial guess is that some box will be required (probably one provided by law enforcement, or, one that meets technical calea specs that must be purchased and installed) that accepts official and secure calea transactions (from law enforcement), and forwards those requested tranactions to asterisk in some form or another. To further advance that guess, calea transactions may simply request certain cdr detail and/or might involve setting up a real-time call monitoring function forwarding audio to the requested calea agency. There is likely some sort of internal logging and reporting function that can be used as a form of checks and balances in subsequent court cases, etc. If those guesses are anywhere near realistic, then I'd further guess that some asterisk app would need to be written to handle at least a portion of the calea tranactions. Anyone care to confirm or elaborate on those thoughts / guesses? ___ --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] CALEA support within asterisk?
Matthew Thompson wrote: On 3 Oct 2006, at 19:53, Colin Anderson wrote: I, for one, welcome our new Republican overlords. lol you are just full of pop culture references, aren't you? Abortions for some, miniature American flags for others. Seriously though - is anyone aware of a precis of CALEA? I'm about to install an Asterisk setup into our US office and being a Brit in the UK I'm not totally up on what your Republic overlords are upto. If you're simply installing asterisk as a pbx in your US office, you probably don't have to do anything (for calea). If you are routing voip calls in a manner more closely resembling the functions of a pstn central office switch where calls through the asterisk system can be to/from anyone (eg, outside your business), then you probably need to keep a close eye on this thread. As one of the other gentlemen on this list suggested, calls originated from your UK office passing through your US asterisk box to your customers (or potential customers) are not governed by calea laws as the calea function would be implemented by the US telco in this case. The exact same applies in the reverse direction. Regardless of which direction the call is flowing in this case, the telco provides the necessary calea functions. ___ --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] T1 timing errors (Frame Slips) on Nortel 61C to TE110P
Ronnie Jones wrote: I am setting up an asterisk box , my first with PRI T1 interface to a Nortel 61C. We have quite a bit of experience with the 61C and do most of the programming including maintaining several other PRI interfaces in this switch. The problem we are having is as soon as we turn up the PRI, on the 61C side we get PRI0264 protocol errors. Then the circuit lays down properly. At this point we start accumulating SLIPR in the 61C which resets the circuit in about 5 minutes. Below are my configurations. Nortel 61C CEQU MPED 8D TERM REMO TERD REMD TERQ REMQ SUPL 004 012 024 V048 N156 SUPC SUPF XCT 000 016 TDS * 000 * 016 CONF * 001 * 017 MFSD * 000 * 016 DLOP NUM DCH FRM TMDI LCMT YALM TRSH TRK 009 12 ESF NO B8S FDL 00 PRI 008 24 ESF NO B8S FDL 00 010 24 ESF NO B8S FDL 00 011 24 ESF NO B8S FDL 00 018 24 ESF NO B8S FDL 00 019 24 ESF NO B8S FDL 00 020 24 ESF NO B8S FDL 00 021 24 ESF NO B8S FDL 00 030 24 ESF NO B8S FDL 00 031 24 ESF NO B8S FDL 00 Blah..blah ADAN DCH 50 CTYP MSDL DNUM 10 PORT 1 DES ippbx USR PRI DCHL 8 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 1 RCAP ND2 MBGA NO OVLR NO OVLS NO T200 3 T203 10 N200 3 N201 260 K7 TYPE RDB CUST 00 ROUT 97 DES IPPBX TKTP TIE NPID_TBL_NUM 0 ESN NO CNVT NO SAT NO RCLS INT VTRK NO DTRK YES BRIP NO DGTP PRI ISDN YES MODE PRA IFC ESS5 SBN NO PNI 1 SRVC NNSF NCNA YES NCRD YES CHTY BCH CTYP UKWN INAC YES ISAR NO CPUB OFF DAPC NO BCOT 0 DSEL VOD PTYP PRI AUTO NO DNIS NO DCDR NO ICOG IAO SRCH LIN TRMB YES STEP ACOD 7997 TCPP NO PII NO TARG CLEN 1 BILN NO OABS INST IDC NO DCNO 0 * NDNO 0 DEXT NO ANTK SIGO STD ICIS YES TIMR ICF 512 OGF 512 EOD 13952 NRD 10112 DDL 70 ODT 4096 RGV 640 GRD 896 SFB 3 NBS 2048 PAGE 002 NBL 4096 IENB 5 TFD 0 VSS 0 VGD 6 DRNG NO CDR NO VRAT NO MUS NO RACD NO FRL 0 0 FRL 1 0 FRL 2 0 FRL 3 0 FRL 4 0 FRL 5 0 FRL 6 0 FRL 7 0 OHQ NO OHQT 00 CBQ NO AUTH NO TDET NO TTBL 0 ATAN NO PLEV 2 ALRM NO ART 0 SGRP 0 AACR NO zapata.conf [trunkgroups] [channels] language=en context=default switchtype=5ess signalling=pri_net usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 musiconhold=default group = 1 channel = 1-23 zaptel.conf span = 1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us I think the configuration is right. I have tried changing the timimg source in zaptel.conf from 1 to 0 to no avail. Also I can not set up the Nortel PRI to look internal for clock. Nortel sets up by default CLOK = EXT. I have tried different cross over cables. I can point the asterisk into a T-Berd 950N set up to turn up a PRI and it will work and run clean on the Asterisk server. I can loop back the Nortel PRI and it will ‘est wrong mode’ and accumulate no SLIPR. I am struggling to get this to work. The circuit does establish and pass calls but resets frequently due to slips. Dell 2850/TE110P/Asterisk business edition ABE-B.1-1/Redhat EL4/Nortel 61C/Succession R3/MSDL Dchannel/NT5D12. Any help would be appreciated. I have no experience on the Nortel side, but will comment on the timing thingie. The asterisk T1 card (port going to the Nortel) will always generate T1 timing on the transmit side of the T1. There is no way to turn it off (by T1 Spec's). So, letting the Nortel use CLOK = EXT is perfect. The sync parameter in /etc/zaptel.conf for that same T1 port should probably be set to zero, but that statement is somewhat dependent on what the other ports on the Asterisk T1 card are used for. If there are no other Asterisk T1 card ports in use, then I'd suggest setting the sync parameter to 1. If at least one other Asterisk T1 port is in use and goes to a central office, then turn that port's sync to 1 and the Nortel port sync to 0. (Keep in mind the digium T1 cards only have one clock on board, and syncing that clock to a T1 coming from a central office is the right thing to do. Once that clock is in sync, then the Nortel will sync to asterisk.) I'm a little confused with your last paragraph when you say the circuit does establish and pass calls but resets frequently due to slips. Are those calls to/from asterisk talking to the Nortel? Or, are you routing incoming pstn calls from the central office through asterisk to the Nortel? Also, have you tried any of the pri show ... commands in asterisk, or any of the pri debug items?
Re: [asterisk-users] Right way to prevent analog channel from answering the phone?
Nick Ellson wrote: I am in the process of learning my A1200P, and i would like an elegant way to prevent it from answering the phone, but still make outbound calls. I tried zap destroy channel 1 (which worked, but pissed off Asterisk ;) Is there a more elegant way to tell it to answer/not answer on command? I don't have an A1200P, but most zap channel interfaces are built to not answer an incoming call unless you specifically configure asterisk to do it. There are only two basic conditions under which an incoming call will be answered: 1. by including the answer statement, like: exten = 3556,1,Answer exten = 3556,2,Wait,1 exten = 3556,3,Authenticate(3017) exten = 3556,4,Meetme(3556|pM) 2. a SIP phone (or other phone) user picks up the handset. So, in zapata.conf you have definitions for each of the A1200P ports, and one of the items in those definitions is context=something. If that context statement points to some non-existent context name (like context=xyz), there is nothing that would answer the incoming call. If the context=something points to a real context (in extensions.conf), then review that context to ensure there is nothing there to answer the incoming call. (Note: some asterisk applications will automatically answer incoming calls.) You could also define that context and include statements like: [no-answer] exten = _X.,1,Hangup ___ --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] How can I stop lost DNS from killing Asterisk?
Eric ManxPower Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? Using IP addresses only does not fix the problem as the asterisk system does not know who he is. Need to define him in /etc/hosts as well, then it works just fine. ___ --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: Setting QOS settings in asterisk and/or CentOS?
Steven wrote: I found this command if your Cisco switches support it: auto qos voip trust You set this on each interface. It automatically prioritizes all SIP and skinny traffic, but not iax. There is also auto qos voip cisco-phone. This one can detect a Cisco phone and prioritize it. I just have to figure out how to verify that it is actually doing anything. The auto qos function is a relatively new addition to the cisco routers and switches (eg, last year or so). The parameter is added to an individual interface (usually a serial interface), and it truly watches for actual traffic on that interface until you shut it down. At that point, auto qos writes the policy statements into the router config needed to support that actual traffic. To use it, you must: - enable it on an individual interface, - do not change the interface bandwidth statement while its running, - cisco express forwarding must be enabled, and, - all previously attached QoS policies must be removed from the interface being sampled. Its my understanding (although I've not actually done this) that auto qos can be used to monitor all traffic and not just voip packets. For example, some companies may wish to generate qos policies for Citrix, MS Terminal Server traffic, etc, and may not have any voip implementation at all. So, auto qos is not just for voip traffic and should be very usable with iax. Since you've specifically mentioned the auto qos voip cisco-phone statement, that statement essentially says watch for voip traffic coming from a cisco phone. Reading between the lines says: Cisco ships their voip phones with QoS already preconfigured with signaling traffic in one DSCP class and rtp traffic in another DSCP class. If your non-cisco phones aren't set up with those exact same DSCP markings, auto qos won't write the policy statements into your router's config. (E.g., cisco tends to push their proprietary voip sutff, so guess what... auto qos voip cisco-phone was oriented around those phones and not necessarily the sip versions of that same cisco phone.) The simplest command is auto qos applied to an individual interface without any other qualifying parameters. Keep in mind that auto qos is actually monitoring your traffic in real time, which assumes you've got voip phones, asterisk box, etc, already preconfigured to mark packets with TOS or DSCP bits. If that's not the case, then your voip traffic appears as default non-qos traffic and no policy will be written to the router's config. For testing purposes, auto qos can be applied to an interface then multiple voip test calls can be initiated manually. It would then write the appropriate policy statements into your config based on those voip test calls. In a large production world, one would apply auto qos to an interface and let it be for some much longer period of time (eg, hours). Then auto qos would write the config statements necessary to support the actual traffic observed over that period of time. There is no magic behind using auto qos; you can do the exact same thing manually by configuring policies in the router and doing something like show policy-map interface s1. That display will tell you how much bandwidth is consumed for each QoS class that has been configured in your policy. The problem with doing that manually is that you have to know when your peak traffic period is for voip traffic, and then run the commands during that peak period to get it right. There are technical white papers on the cisco web site (somewhere) that describes how to use the auto qos function, but keep in mind the function was only recently introduced so it is not yet implemented on every product or in every IOS image. ___ --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] How can I stop lost DNS from killing Asterisk?
Eric Bishop wrote: Hi All, When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? We just went through the same problem. You need both a caching dns server, and, define your asterisk system in /etc/hosts so he knows who he is. I've tested this several times as we use a laptop to demo asterisk and several of these demo's don't have any internet access. (And, you're right, asterisk does not process any calls.) With dns caching and the /etc/hosts definition in place, it now works everywhere. ___ --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] Line Pickup Problem
Pato Valarezo wrote: Lacy Moore - Aspendora wrote: Wherever you have your exten = s,1,Answer statement, replace with: exten = s,1,Wait(30) ; or however long you want to wait to give someone else the chance to answer exten = s,n,Answer then continue on. Asterisk will then wait 30 seconds before it answers the phone. You would probably want this a lower number, though. Hi, i'm using x100P clones and i have two related issues: 1. In the first system (or in both) when someone answer the call, asterisk doesn't notice the stop ringing signal and continues with the dialplan, and of course answer the call and plays the welcome message and interrupts the current call in progress. 2. One of the system wich is connected to the PSTN doesn't seems to wait the time i specify in exten = s,1,Wait(10), and answers the line in a shorter time... it seems like the time doesn't count to it. I'm testing and training with this systems until i can buy a better quality hardware i expect to not have this problems with digium or better hardware. If someone has experience in this i'll apreciate comments. Based only on the words that you've used above, it sounds like you have a problem with extensions.conf (and maybe with the 'context' associated with the x100p card. To better understand your issue, we'll need to see your extensions.conf file and zapata.conf file contents. I'd suggest not trying to copy/paste a piece of those two files but rather include the entire files. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Yes, have multiple clients with asterisk behind a sonicwall. I don't understand from your wording if you mean a voip connection suddenly changed from dup/5060, or, did you change the asterisk system to use some other udp port. The sonicwall does have an option to support sip (udp/5060), but I've not had to use it on anything that we've worked with. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no real connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Are you sure that you're not confusing the SOURCE vs DESTINATION port? Your system would send a sip packet to your provider with a destination port of udp/5060, but your source port can be anything greater then 1024. (That's likely to be 36612 in your notes, above.) Your provider would reverse those in its response, sending their packet to the destination port of udp/ (the same greater then 1024 mentioned above), and a source port of udp/5060. That's just standard IP stuff. The nat function within the firewall keeps track of every udp and tcp conversation by building a table entry that includes source IP and source port (associated with the internal lan device that created the packet), and a destination IP and destination port (associated with your provider's device. That table entry is constantly referred to for every packet that passes through the firewall in either direction, translating private addresses into public addresses, etc. If the conversation is udp based, that table entry will timeout (and disappear) after some period of time. I don't recall what the default sonicwall timeout value happens to be, but its typically some number of low minutes (as opposed to low number of seconds). If the conversation is tcp based, that table entry will disappear when the tcp session is closed by the end devices. I can only guess that a tcp timeout value exists as well, however it would oriented around timing out a table entry where the end devices mysteriously disappeared (without closing the tcp session). Sonicwall sells their products with 10 user, 25 user, and other limits that would imply the above nat table size might have limits (or changes) when that maximum is reached. Are you sure you've not exceeded the license limit associated with your sonicswall? Sonicwall also has a problem handling udp packets that are greater in size then 1458 bytes (I think I have that value correct) when its wan interface is configured for PPPoE. Packets larger then that value are simply dropped on the floor, and no log entries are created to hint that has happened. Are you using PPPoE? Finally, sonicwall has implemented some sort of sip fixup that attempts to analyze the contents of a sip packet to determine which udp ports are to be used for rtp packets. I wouldn't think this function would have any impact in your case since it sounds like the problem is sip oriented and not rtp oriented. You could turn that option off just to ensure it isn't the problem. To diagnose this any further really requires a packet sniff (eg, ethereal) from the outside edge of the firewall, along with an asterisk 'sip debug'. That would help determine what might be happening in terms of port mapping, etc. ___ --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: Very high ping times from 7960 phones
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cisco 7960 phones report times 150ms. Every single one. I've scoured the settings on the 7960's and have looked and looked for why this might be the case. Cisco ata's (186) on the same network report ~ 10 ms. An xlite softphone reports ~ 5ms regardless of what computer it's installed on. I have two 7960 phones with 7.4 firmware and sip show peers tells me that response time is 70 and 72 ms. Hope this helps. I can't tell you why either, but a ping from a linux command line shows sub-millisecond response (phone and asterisk on same lan segment), while the qualify response time is around 79 milliseconds. Just taking a pure guess (without doing any packet sniffing) is the qualify method sends a sip packet to the phone and waits for a response. It is entirely possible that qualify ping might involve multiple packet interactions. Also, the qualify ping must essentially pass through all of the asterisk code, IP stack, etc, on both ends. That value would be greater then a simple icmp ping. There are no settings in the cisco phones that would impact this. ___ --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] Cisco 7970 - DTMF
Tomislav Parčina wrote: In sip.conf for one friend (Cisco 7970 phone) I have define this dtmfmode=inband And in xml.conf of that phone I have preferredCodecnone/preferredCodec dtmfAvtPayload101/dtmfAvtPayload dtmfDbLevel3/dtmfDbLevel dtmfOutofBandnone/dtmfOutofBand But DTMF doesn't work for that phone. Phone establishes call using g711 alaw codec. How should I configure phone and sip.conf to make DTMF work? In asterisk sip.conf, use dtmfmode=rfc2833 for that extension, and in the SIPDefault.cnf boot file for the cisco, include: dtmf_inband: 1 dtmf_outofband: avt dtmf_db_level: 3 (you'll need to translate the above 7960 parameters into the 7970 xml parameters since I don't have a 7970 to play with.) Taking a wild-ass guess, you might be able to get by simply using the dtmfmode=rfc2833 parameter in asterisk without touching the phone. Try it. ___ --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] voicemail greeting
unplug wrote: Hi, When I use Voicemail function, there is a default system greeting before voicemail recording. Is it possible to change that greeting? How? Call into voicemail as though you were going to listen to your messages, and press 0 for Mailbox Options. Then press 3 to record your name. You might want to go through each of the various voicemail options to see what else you might be missing. There are more options. ___ --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] TDM2400P vs Sangoma A200
I'm putting together a plan for a new Asterisk system and I'm trying to decided on an interface card to use. I was originally planning on using a Sangoma A200 but now I'm considering a Digium TDM2400P. The server is large enough to accommodate the full sized TDM and I'll be using 8 FXO channels so molex power connectors aren't an issue. The connector will be slightly more to deal with but not a biggie. Either card I get will have the on-board echo canceler. For the extra $150 for the TDM, not having to mess with two sets of drivers is pretty appealing. Anyone have experience with both cards to give advice one way or the other? (And in case anyone suggests I just go with a PRI, I can't. I'm stuck with POTS lines for now). Over the past three years, I've used the x100p, tdm04b, and the a200d. I've got a tdm2400 here (on loan), however due to personal issues (cancer) I'll need to return it without doing a formal comparison. My inclination would be to go with the a200d as sangoma seems to fix issues rather quickly. If you check the archives for the tdm2400, you'll see where fixing issues with it have drug out over months and I'm not sure if all the known issues have truly been addressed as yet. In addition, the tdm2400 card uses the large centronix-looking connector, so you'll need to purchase a patchpanel (or whatever) to break those 24 ports out into something usable. The a200d has been in use here for about six months and has provided rock solid performance over that time. I've not even bothered to upgrade the drivers for it. I'm using both the fxs and fxo modules, and do send fax calls via the card. ___ --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] SPA941 - Asterisk - Voip provider - PSTN - ShoreTel garble
Cliff Brake wrote: I am using the following setup: Linksys SPA941 - Asterisk - NuFone - PSTN - ShoreTel system The system works great for the most part. Most people I call say it sounds good. However, every time I call a certain company that uses a ShoreTel system, they claim the sound is garbled (understandable, but not pleasant to listen to). Everything sounds fine at my end. If I make a call w/ the following setup, it sounds fine: Analog phone - Asterisk:TDM400 - NuFone - PSTN - ShoreTel system So, it seems there is some type of weird interaction between my system and the ShoreTel system if I use the SPA941 IP phone. Does anyone have suggestions as to how I can start debugging this? Check the RTP Packet Size (under the Sip tab). Set it to .020 (20 milliseconds) and place another test call. For whatever reason, the Linksys/Sipura products default to 30 milliseconds and has impacted the quality of audio on some systems. ___ --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] Setting QOS settings in asterisk and/or CentOS?
BerkHolz, Steven wrote: How would I go about setting the TOS bit to RTP IP TOS Byte: 18 (hex) for SIP and IAX traffic at the asterisk server? Also, Do you have a quick reference on how to configure a Cisco switch to prioritize SIP traffic? I check in various Cisco docs, and there are so many references, and none of them seem to relate directly to using the TOS bit for QOS. I am looking into using the TOS bit because that is the only method that my SIP devices use. (Citel Handset Gateway) For asterisk, take a look at sip.conf.sample and you'll find something like this for v1.2 and earlier: tos=lowdelay;ox18 sets ip tos bits (=lowdelay, throughput) After v1.2, the look in the asterisk/docs directory and you'll find a readme file relative to QoS. The format of the QoS parameters have changed from the older TOS bits to the newer terminology Differentiated Services, and coding within sip.conf looks something like this: tos_sip=cs3 tos_audio=ef Differentiated Services is a superset of TOS; anything you want to do in TOS bits have an equivalent in Differentiated Services, and the bits map exactly. The cisco web site has a very significant amount of documentation for configuring routers and switches for QoS, and they have a very excellent 700+ page book that is oriented 100% towards implementing QoS on various cisco boxes. Cisco's search engine leaves something to be desired in some cases, but the info you want is there. Not all cisco switches have the same QoS implementations. For example, most of the workgroup type switches support something like 3 or 4 outbound queues, while the higher end switches support more queues. If you're going to deal with RTP only from a QoS perspective, you only need two queues (eg, RTP Default). The Default queue (or Class) is a special case that includes everything not in other queues. For the most part, QoS on switches is not required unless: a) trunk port traffic exceeds the bandwidth available (for that port), or, b) outbound port is a slower speed then the majority of other switch ports (eg, congestion). ___ --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] Setting QOS settings in asterisk and/or CentOS?
Nick Hoffman wrote: On Sat September 23 2006 06:14, Bob Amen [EMAIL PROTECTED] wrote: snip which sets the TOS bit on all IAX, SIP and RTP packets. Using iptables means that we can set up our rules on the router without using ACLs. Our Cisco Cookbook (http://www.oreilly.com/catalog/ciscockbk/) has a nice section on QoS (Chapter 11) and an appendix on TOS, etc. The author advises not to use ACLs when possible as they take more CPU in the router to implement and on a heavily loaded router can cause packet delays. So here's what our config looks like: snip Cheers, Bob Hi Bob. I'm new to TOS and DSCP, but after going over your and Rich Adamson's responses to Steve BerkHolz's question, I read up about them. With what you wrote above, does this mean that your Cisco router(s) deny, allow, and route traffic based on TOS/DSCP flags, and you don't bother with traditional ACL rules like below?: access-list 123 permit udp 1.2.3.4 ... ACL's in cisco hardware can be used for pattern matching in addition to the old permit, deny, etc, functions. Here's a working example from a cisco 1750 with QoS: class-map match-all voice-rtp match access-group 103 class-map match-all www-traffic match access-group 105 ! ! policy-map voice-policy class voice-rtp priority percent 40 class www-traffic bandwidth percent 30 class class-default fair-queue access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www any In the above, any packet matching the access-list 103 gets treated as a voice-rtp class, and in the policy map, is acted upon as priority (which means low latency queue) and can use up to 40% of the interfaces bandwidth. The bandwidth 384 statement on the interface is used by QoS to determine how much is actually going to be used for voip. interface Dialer0 bandwidth 384 ip address negotiated encapsulation ppp dialer pool 1 dialer-group 1 service-policy output voice-policy ppp pap sent-username x_dsl password 7 136775499987 That bandwidth statement should be the actual amount of bandwidth available and not the value that your dsl/broadband provider says they provide. Once the policy map is implemented, one can review the operational statistics by doing something like this: C1750#show policy-map interface dialer0 Dialer0 Service-policy output: voice-policy Class-map: voice-rtp (match-all) 1441504 packets, 191386680 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 103 Weighted Fair Queueing Strict Priority Output Queue: Conversation 136 Bandwidth 40 (%) Bandwidth 153 (kbps) Burst 3825 (Bytes) (pkts matched/bytes matched) 0/0 (total drops/bytes drops) 0/0 Class-map: www-traffic (match-all) 484061 packets, 341420115 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: access-group 105 Weighted Fair Queueing Output Queue: Conversation 137 Bandwidth 30 (%) Also, by doing the following: C1750#show access-list 103 Extended IP access list 103 permit ip any any dscp cs3 permit ip any any dscp ef (1680 matches) permit ip any any tos min-delay (808709 matches) permit ip any any tos 12 (1 match) one can see which piece of an access list is being matched. One can also see that both TOS and DSCP definitions can be used within the same access list. Its kind of a handy way to ensure voip phones and asterisk are properly configure and thus properly treated from a QoS perspective. It should also be noted the above router is running v 12.2(4)T7 code. Cisco has made several changes to the syntax and parameters implemented in each version in the last few years. In the newer IOS versions (for both switches and routers), the syntax and parameters are becoming much more standardized across all product lines. The OP was specifically asking about QoS on a cisco switch, and without researching exactly what was implemented in his switch, there really isn't any way to give him a QoS template that would be accurate. For example, if I posted something that worked in the 12.4 code, its highly likely not to be acceptable syntax for 12.1 or 12.2. Whether one uses access lists to do pattern matching is mostly immaterial except on a heavily loaded router. In my case, the processor utilization looks like: C1750#show proc CPU utilization for five seconds: 1%/0%; one minute: 1%; five minutes: 1% where lengthy access lists would have almost zero impact. For those that have read this far, it should be noted the implementation is a 3-queue policy (one for rtp, one for www, and one as the default). If the traffic for the rtp queue is low (or none), the unused bandwidth is automatically made available to other lower priority queues. In other words, the
Re: [asterisk-users] Asterisk Design Question
Remi Quezada wrote: Hi, Right now I am in the process of setting up an asterisk box. I was thinking of having two asterisk box, one that is hooked up to the PSTN using a digium TE405P card and the other asterisk box will be used to store all the sip user features and routing information. Do you think this a good design? Or do you think I should just stick with having one asterisk box that does everything. I plan on having a lot of users hooked up to it in the future. The system specs are 3.0 GHz Pentium 4, 1 GB RAM, and a 40 GB hard drive. I attended a cisco presentation a while back and they indicated the architecture of their system was changing somewhat (away from Windows, now on Linux, etc). The presentation suggested that certain functions are dedicated to certain systems/boxes, and if one needed more of a certain function then add another box. For example, if transcoding is a requirement, then dedicated a box or two to that function. As the overall system grows and more transcoding is needed, add another box for that. Since I'm not a cisco reseller, etc, I didn't keep very many notes relative to the above. But, the approach seems to be one that can support long term growth is small increments of hardware/software. Your approach kind of follows cisco's in a way. The only issue (from a high level) that might be difficult to handle is that asterisk really wasn't designed to distribute functions to multiple boxes. E.g., if growth dictated two pstn interface boxes, how does one manage the distribution of pstn calls from a single routing box (including pstn T1 failures, overloads, etc)? ___ --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] Why not g726-32?
RR wrote: On 9/16/06, Rich Adamson [EMAIL PROTECTED] wrote: RR wrote: All, is there anyone who uses g726-32 ? If not, then does anyone know why don't people use it? I use g726 on iax links between systems and to teliax.com for LD calls. Have no idea if its -32 or what though. What ships with asterisk (in terms of g726) has been working very well for us with the exception of a period of time where all g726 calls via teliax were not usable. Teliax had to have had a problem or was playing around as that was the only iax link that had bad audio. Thanks Rich for the positive email about g726. Just FYI, (*) supports only g726-32 AFAIK so that's probably what you've been using. I don't have the worry of Teliax as I'd probably never be using them or at least not in the immediate/near future. Like I said, all I want to do is avoid usage of license fees, save bandwidth, and not stress out my systems with cpu intensive codecs like g729 and maybe find something that can still deliver comparable quality. I'm still confused as to why more people and carriers don't use g726 however. I can only guess that many itsp's actually support it, but don't advertise its availability, just like they don't advertise ilbc, etc. I'd also have to guess that phone manufacturers haven't implemented it (in the past) due to limits on memory, etc. ___ --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] Cisco Distinctive ring using alert-info
There has been several different hardware versions of the phone, but to the best of my knowledge, the ringer has not changed. The cisco documentation suggests there is a way to create your own ring tones, but I've not tried that either. The stock 7960 sip phone's built in ring tones are not very impressive, and as I recall, are basically limited to sounds such as one-long, one-long one short, etc. Lacy Moore - Aspendora wrote: Do some 7960s perform differently? On 9/15/06, *Eric ManxPower Wieling* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Rich Adamson wrote: Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a normal ringtone, with foghorn being used if the call was coming in from the girlfriend/wife/mother-in-law etc ;) I was trying to use the following: exten = 5711,1,SIPAddHeader(Alert-Info: Foghorn) exten = 5711,n,Dial(SIP/5711) exten = 5711,n,Hangup() However, not matter what I try, I get the standard ringtone. If I use exten = 5711,1,SIPAddHeader(Alert-Info: Bellcore-dr3) Our past experience indicates the Bellcore-dr3 approach is the only one that works. That TOTALLY depends on the phone. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- 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 -- Lacy Moore Aspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Distinctive ring using alert-info
Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a normal ringtone, with foghorn being used if the call was coming in from the girlfriend/wife/mother-in-law etc ;) I was trying to use the following: exten = 5711,1,SIPAddHeader(Alert-Info: Foghorn) exten = 5711,n,Dial(SIP/5711) exten = 5711,n,Hangup() However, not matter what I try, I get the standard ringtone. If I use exten = 5711,1,SIPAddHeader(Alert-Info: Bellcore-dr3) Our past experience indicates the Bellcore-dr3 approach is the only one that works. ___ --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] Cisco Distinctive ring using alert-info
Eric ManxPower Wieling wrote: Rich Adamson wrote: Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone. However, I was wanting to use a normal ringtone, with foghorn being used if the call was coming in from the girlfriend/wife/mother-in-law etc ;) I was trying to use the following: exten = 5711,1,SIPAddHeader(Alert-Info: Foghorn) exten = 5711,n,Dial(SIP/5711) exten = 5711,n,Hangup() However, not matter what I try, I get the standard ringtone. If I use exten = 5711,1,SIPAddHeader(Alert-Info: Bellcore-dr3) Our past experience indicates the Bellcore-dr3 approach is the only one that works. That TOTALLY depends on the phone. Which he mentions is a cisco 7960, and its sip image does the Bellcore thingie. ___ --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] Why not g726-32?
RR wrote: All, is there anyone who uses g726-32 ? If not, then does anyone know why don't people use it? I use g726 on iax links between systems and to teliax.com for LD calls. Have no idea if its -32 or what though. What ships with asterisk (in terms of g726) has been working very well for us with the exception of a period of time where all g726 calls via teliax were not usable. Teliax had to have had a problem or was playing around as that was the only iax link that had bad audio. Oh and since I am only looking at codecs to use between the subscriber and our system (no carriers involved), the popularity and ubiquity of g729 and g711 aren't a qualifying factor for this particular discussion :) ___ --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] Reliability of the newer IAXy's
The sipura stuff (and lots of other ata's) work just fine behind most nat boxes if the asterisk box is on a registered IP. John Novack wrote: Regarding the IAXy, newer model- S101i I have an application for one. Both the IAXy and the Asterisk would be behind routers ( cheap Linksys ones ) , both ends with a dynamic ( subject to change ) IP address. I have RTFM, such as it is, and really don't see how it can be properly configured Would I be better off with the 2100 and tough through the NAT issues? Any suggestions?? John Novack Andrew Joakimsen wrote: Honestly for the price its a bad unit. If they were priced $40-50 then yes its a great unit. But at $90, single port, no web config, very basic provisioning and with the S100 we had many issues of reliablity where a SIP ATA did not have the same issues. Maybe it's heat, but thats an issue I would expect from a $40 chinese ATA, not something that cost over $100 at the time. For $10 less you can find the SPA-2100 which has double the voice ports, a built in router and more options. On 9/15/06, Lists [EMAIL PROTECTED] wrote: Is anyone out there currently using the newest model IAXy? I was thinking about purchasing one for testing but was wondering if they have gotten any better than the original models. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 9 becomes 99 ? And other strangeness
Brian Candler wrote:
I'm getting a strange situation with the first digit being doubled on
outbound dialling, and other oddities. I think something strange is going on
in my dialplan, rather than a DTMF decoding issue, but see what you think.
The platform is CentOS 4.4 plus Asterisk SVN trunk as of yesterday, and a
TDM400P with 2 x FXS and 1 x FXO.
Here's my extensions.conf, based heavily on the simple examples from the
O'Reilly Starfish book:
---
[outbound]
exten = _9.,1,Dial(Zap/4/${EXTEN:1}) NOTE HERE
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()
Try replacing the first step above with:
exten = _9.,1,Dial(Zap/4/w${EXTEN:1})
Note the w in the above means wait for about a 1/4 second before
sending the number to the central office.
Some central offices are not ready to receive digits as quickly as
asterisk sends them out. In fact, some users have to use multiple w's
(as in Zap/4/www${EXTEN:1} to wait for the equipment to settle down.
Give that a try and let us know if it corrected the problem.
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Re: [asterisk-users] 9 becomes 99 ? And other strangeness
Brian Candler wrote:
On Thu, Sep 14, 2006 at 09:00:57AM -0500, Rich Adamson wrote:
[outbound]
exten = _9.,1,Dial(Zap/4/${EXTEN:1}) NOTE HERE
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()
Try replacing the first step above with:
exten = _9.,1,Dial(Zap/4/w${EXTEN:1})
Note the w in the above means wait for about a 1/4 second before
sending the number to the central office.
Some central offices are not ready to receive digits as quickly as
asterisk sends them out.
Interesting feature, thank you, but I don't think that's the problem.
Notice that Asterisk's own log shows that it thinks the number called is
99X and therefore dials out to 9X, where in fact I only dialled
9X and so it should be dialling X.
Console:
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1,Zap/4/907974XX) in
new stack
-- Called 4/907974XX
-- Zap/4-1 answered Zap/1-1
-- Native bridging Zap/1-1 and Zap/4-1
-- Hungup 'Zap/4-1'
== Spawn extension (internal, 9907974XX, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
If this were consistent I could use ${EXTEN:2} to strip off the two 9's, but
it isn't.
Try the above an see what the result is. If it does not address the
problem, at least one item has been removed from the list of
possibilities. ;)
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Re: [asterisk-users] question...
If you have four pstn telephone numbers (eg, 444-1212, 444-1213, 444-1214, and 444-1215) from your telco, then call the telco and have them implement call forwarding on each of the four lines. You might also verify they provide a call forwarding on busy function for those lines. After they have implemented it, put an analog phone on line 444-1212 and implement call forwarding on busy using whatever codes are appropriate (*90 here), forwarding calls to 444-1213. Do the same for 444-1213 and 444-1214. Now when 444-1212 is busy, the next incoming call goes to 444-1213. When 444-1213 is busy, the next call goes to 444-1214, etc. Christopher Corn wrote: rich, thanks for replying. i assume your talking about enabling call forward and call forward on busy from my vsp side. i dont quite grasp everything else that your saying, can you explain in laymen terms. thanks. */Rich Adamson [EMAIL PROTECTED]/* wrote: Christopher Corn wrote: i plan on buying 4 residential lines for our small office and i was giving some thought. we'd like to have one main number that can transfer calls to the other lines. but seeing that i have 4 different individual lines with different numbers, im not seeing hows thats possible, without tying up a line on the main phone. i would think i would need one DID with multiple simultaneous connections. Two ways to accomplish the objective. 1. ask the telco about four lines in a trunk group (or sometimes referred to as a rotary hunt group). 2. Subscribe to call forwarding on each line, and program each line for call forward on busy to the next line of the four. It will accomplish the same thing as the trunk group approach above. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI: sometimes Asterisk drop calls
Steve Davies wrote: For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo If memory serves correctly, I believe the parameter was added a couple of years ago as a means / workaround for hung channels. At the time, there was not any overwhelming evidence as why a channel would occasionally hang. Some of the possibilities included unusual interaction from the opposite end of the T1/E1, anomalies in the dialplan, etc. Now that a substantial amount of work / changes have been made relative to PRI's and other internal asterisk code, there appears to be less of a need to reset. A reasonable approach might be to apply the parameter and pay close attention to channels that might be in some strange state. If none are observed, then leave it. ___ --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] PRI: sometimes Asterisk drop calls
Steve Davies wrote: On 9/12/06, Rich Adamson [EMAIL PROTECTED] wrote: Steve Davies wrote: For the curious, can anyone tell me how this flag fixes the issue? - I have seen the error before, but always assumed it was related to hung channels. Thanks, Steve On 9/12/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Problema solved! Just put resetinterval=never inside zapata.conf Giorgio Incantalupo If memory serves correctly, I believe the parameter was added a couple of years ago as a means / workaround for hung channels. At the time, there was not any overwhelming evidence as why a channel would occasionally hang. Some of the possibilities included unusual interaction from the opposite end of the T1/E1, anomalies in the dialplan, etc. Now that a substantial amount of work / changes have been made relative to PRI's and other internal asterisk code, there appears to be less of a need to reset. A reasonable approach might be to apply the parameter and pay close attention to channels that might be in some strange state. If none are observed, then leave it. Thanks for that. I have a customer who is using Asterisk 1.0.x, and I am tempted to backport this fix from the 1.2.x code where it was introduced. From a personal perspective, I think I'd hold off on the back port and devote that time towards testing the soon to be released version (now in Trunk). If you've watched the number and type of changes that have gone into SVN Trunk in the last couple of months, it appears as though a significant number of possible memory leaks, sip code, infrastructure code, PRI code changes, etc, have been applied that would be beneficial for all production systems. There also appears to be a fair amount of work that will be needed to upgrade dialplan syntax (etc) for the new release. Best guess is that once the Trunk code gets past the beta testing phase, it will likely be the asterisk code of choice for most/all production systems. Consider the above is only my $0.02 worth. ;) ___ --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] Switch Experiences
Ben Gore wrote: Hello: I'm would like to get feedback before finalizing design of a VOIP network, in particular about people's experience with network (primarily 10/100/1000 twisted pair) ethernet switches. I have a number of candidates in mind, but I would like any and all opinions and suggestions on the following topics: -Throughput/minimal latency/delays; -Managed vs unmanaged; -Redundant links/auto healing; -Redundant power supply; -Configuration of port attributes (i.e. locking 10 M/b interface to 10 M/b instead of leaving in AUTO); -Resistance to Electrostatic/Electromagnetic/RF energy; -Shielded vs unshielded ports cables; -Pricing; -Any other relevant information. The reason for asking is there seems to be a significant amount of disagreement about a number of these issues from a variety of experts, while there's a considerable amount of experience on this list in these areas. Suggestions of specific manufacturers and models welcome if you've had good luck with them. What you're asking is highly dependent on the size of network. I've been doing network performance analysis and voip readiness assessments as a consultant for companies in 40+ states and have seen about every combination of hardware that exits. Two examples FWIW. A college wanted to implement voip to each of 30+ buildings using their existing flat cisco-based network. The buildings were all interconnected with gig fiber, however each building had workgroup type switches. Since their gig backbone was very much under-utilized, there really wasn't a current need to implement QoS, etc. A key element however was that every switch was of the managed type and the college had facilities in place to monitor dropped packets, etc, for every switch port. They implemented a commercial voip system and are rather happy with its overall operation. The flip side are most small businesses that purchase unmanaged switches from the cheapest supplier they could find, and have no eyes into how their backbone is performing. Although a small voip system might function reasonably well, there is no way to identify disruptions (such as dropped packets). If you intend to purchase switches that support QoS, then dig into exactly how QoS is implemented and you'll narrow your choices rather rapidly. For example, there is a large number of switches that say they support QoS, but in reality they have implemented QoS packet marking on a per-port basis and nothing in terms of queue control (eg, no outbound QoS queues on the ports). From a marketing perspective, they can claim QoS support but you'd never be able to do anything constructive with it. Most all current switches (even the cheapest models) are very reliable (no need for redundant power supplies), transfer packets basically at wire speeds, and are quickly moving towards a commodity item. Also, the majority are built overseas by companies for US manufacturers. Take the cover off any of the well-known brand-name switches and you'll find part numbers and manufacturer's names that are very different then what is on the front panel. (And, many of us understand contracting the build process verses simply purchasing a large lot of pre-manufactured boxes, rebranding, etc.) Since you're asking on this list, you're probably wanting a switch that supports QoS in some manageable form. Those that do manage QoS well generally do all the other things mentioned on your list very well. So, reorient the research into one of identifying those managed switches that have implemented QoS well. ___ --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
I have two contexts how could I isolate context A from context B ,in
other words I want to ban context A from calling context B
In sip.conf, define phones/extensions something like this:
[1000]
type=friend
other parameters as needed
context=cust-a
[1001]
type=friend
other parameters as needed
context=cust-a
[2000]
type=friend
other parameters as needed
context=cust-b
[2001]
type=friend
other parameters as needed
context=cust-b
In extensions.conf, define dialplans something like this:
[cust-a]
include=local-extn-cust-a
include=local-calls-a
include=misc-extns
include=no-match
[cust-b]
include=local-extn-cust-b
include=local-calls-b
include=misc-extns
include=no-match
[local-extn-cust-a]
exten = 1000,1,Dial(SIP/1000,15,r)
exten = 1000,2,Voicemail(1000|ug(6))
exten = 1000,102,Voicemail(1000|bg(6))
exten = 1000,103,Hangup
exten = 1001,1,Dial(SIP/1001,15,r)
exten = 1001,2,Voicemail(1001|ug(6))
exten = 1001,102,Voicemail(1001|bg(6))
exten = 1001,103,Hangup
[local-extn-cust-b]
exten = 2000,1,Dial(SIP/2000,15,r)
exten = 2000,2,Voicemail(2000|ug(6))
exten = 2000,102,Voicemail(2000|bg(6))
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,15,r)
exten = 2001,2,Voicemail(2001|ug(6))
exten = 2001,102,Voicemail(2001|bg(6))
exten = 2001,103,Hangup
[local-calls-a] ; outgoing pstn calls for cust-a
exten = _21X,1,Dial(Zap/g1/${EXTEN})
exten = _30X,1,Dial(Zap/g1/${EXTEN})
etc
[local-calls-b] ; outgoing pstn calls for cust-b
exten = _21X,1,Dial(Zap/g2/${EXTEN})
exten = _30X,1,Dial(Zap/g2/${EXTEN})
etc
[misc-extns]
exten = 3912,1,Wait(1)
exten = 3912,2,SayDigits(${CALLERID(num)})
exten = 3912,3,Hangup
[no-match]
exten = _X.,1,Answer
exten = _X.,2,GotoIF($[${EXTEN} != h]?10)
exten = _X.,10,Playback(invalid,skip)
exten = _X.,11,Hangup
In zapata.conf (assuming you have some zap pstn interfaces for each
customer), use something like this:
context=cust-a
other needed parameters
group=1
channel = 1,2
context=cust-b
other needed parameters
group=2
channel = 3,4
The above is a very simple example. Those extensions belonging to cust-a
cannot call those extension belonging to cust-b, and outgoing pstn calls
from each customer uses zap interfaces belonging to each customer.
If you're using [EMAIL PROTECTED], Trixbox, or some other pre-canned
implementation of asterisk, then pose your questions on their respective
support lists.
___
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Re: [asterisk-users] question...
Christopher Corn wrote: i plan on buying 4 residential lines for our small office and i was giving some thought. we'd like to have one main number that can transfer calls to the other lines. but seeing that i have 4 different individual lines with different numbers, im not seeing hows thats possible, without tying up a line on the main phone. i would think i would need one DID with multiple simultaneous connections. Two ways to accomplish the objective. 1. ask the telco about four lines in a trunk group (or sometimes referred to as a rotary hunt group). 2. Subscribe to call forwarding on each line, and program each line for call forward on busy to the next line of the four. It will accomplish the same thing as the trunk group approach above. ___ --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] Zaptel-1.2.9 compile error
Samy Antoun wrote: --- Bill Maidment [EMAIL PROTECTED] wrote: Hi I've just tried to compile the zaptel-1.2.9 release and I get the following error: Same here, using CentOS 4.4 kernel 2.6.9-42.0.2.ELsmp, got these errors when compiling zap: make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: /usr/src/zaptel/wct4xxp/../oct612x/octasic-helper: Command not found make[3]: *** No rule to make target `/usr/src/zaptel/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h', needed by `/usr/src/zaptel/wct4xxp/vpm450m.o'. Stop. make[2]: *** [/usr/src/zaptel/wct4xxp] Error 2 make[1]: *** [_module_/usr/src/zaptel] Error 2 make: *** [linux26] Error 2 Hope someone has a workaround for this problem Have you tried: cd /usr/src/zaptel make update make install ___ --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] Zaptel-1.2.9 compile error
Bill Maidment wrote: Rich Adamson wrote: Have you tried: cd /usr/src/zaptel make update make install I've never used the tarball, however if the tarball is installed and the resulting code can't be compiled, obviously there is a Makefile present. Part of the Makefile includes update, so make update should work just fine (from whatever directory the source was installed into). The make update will use the svn source which is the exact same source that was used to created the tarball in the first place. So, yes it will work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom related question
Kevin Smith wrote: Hi everyone, While this isn't a true asterisk question, I know a lot of people here use Polycom phones. Anyway, I have two Polycom 601 phones that share the same voicemail box. Now it is intermittent, but sometimes both phones will have a notification there is a voice mail, but then sometimes only one will show that there is a voicemail. If the phone that doesn't show there is a voicemail connects to the voicemail box it can get the message, but just no indication. My question is, has anyone else tried doing this and had success? If so is there anything on Asterisk that I need to set or in the configuration for the phones that I may be overlooking? Without seeing you configs, I'm not sure this will answer your question. If you look at the sample configs, you'll find: [EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple mailboxes in the sip.conf.samples for v1.2 stable. That is the only way that I know of to turn on the mwi for two different phones (eg, extensions). Is that what you're using and its not working? ___ --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] Voip providers and sip origination and termination?
Christopher Corn wrote: can someone please explain the differnces to me??? I have an asterisk system im setting up for a small office (4 or 5 phones) and as im looking for a voip provider, i find that voip providers generally have unlimited plans, and those that offer sip origination and termination get charged for the minute, for their outgoing and incoming calls. is there a difference in the backend architecture here? if so, what? or is this is just a difference in marketing terms and setup? for example, http://www.broadvoice.com offers an unlimited plan in the US for calls, though they never use the term sip origination and termination. they say their systems also supports asterisk. yet http://www.bandwidth.com/content/enterprise?page=voice_services_origination_terminationcampaignId=7013JBJ http://www.bandwidth.com/content/enterprise?page=voice_services_origination_terminationcampaignId=7013JBJ calls it sip origination and termination any info is appreciated! thanks! I'll take a stab at this... There are some providers that allow you to originate calls to the US/World pstn network via their facilities, but do not provide any way for the US/World to call you from the pstn network. (eg, Origination only provider.) There are many providers that do the above, but also will assign you a normal pstn telephone number allowing the US/World pstn users to call you (via sip, iax, etc). (eg, Origination and Termination provider.) The back end differences for the providers essentially amounts to them having to purchase multiple T1's, obtain an allocation of pstn telephone numbers, and establish a dialplan to support calls from the pstn network. The architecture for origination-only verses origination plus termination is the same; the implementation is different for one verses the other. For the most part, there are no providers that truly provide unlimited service. The majority include words in fine print that impose some sort of limit on their so called unlimited service. For example, some will say things like their unlimited service provides 2500 minutes of use; call volumes that exceed 2500 minutes will be billed at $0.02/minute. Got to read the fine print. From an architectural perspective, those providers that suggest they have unlimited service plans also impose a limit on how many simultaneous calls are allowed. The majority of these have a limit of one, two, or some very small number of simultaneous calls. There way of limiting usage since they don't really want you to use up more then their stated fine-print usage. Those providers that sell their services based on a cost per minute (as opposed to unlimited plan) do not typically limit the number of simultaneous calls. They want you to use as many minutes as possible, so why would they try to limit the number of simultaneous calls? To get the best deal possible (from any provider) you need to come up with a reasonably accurate estimate of the number of minutes of incoming and outgoing calls that you are going to make. Then, compare providers to see which ones cost the least in terms of your requirements. Keep in mind the higher your call volumes, the more competitive the providers are. In other words, if your needs suggest 1,000,000 minutes of use per month (incoming and outgoing), you should be able to find providers that will charge you something like $0.012 per minute. (Stated a little differently, the majority of service providers have other unpublished plans that are discounted based on your expected level of usage.) Most providers are trying to pattern their plans based on how well the Cell providers have done in the past. You and I typically sign up for minutes of cell phone usage, but don't actually use all of those minutes. What's our real cost per minute in this case? And, how often do we make useless cell phone calls because we have free minutes left? ___ --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] Zaptel-1.2.9 compile error
Bill Maidment wrote: Rich Adamson wrote: Bill Maidment wrote: Rich Adamson wrote: Have you tried: cd /usr/src/zaptel make update make install I've never used the tarball, however if the tarball is installed and the resulting code can't be compiled, obviously there is a Makefile present. Part of the Makefile includes update, so make update should work just fine (from whatever directory the source was installed into). The make update will use the svn source which is the exact same source that was used to created the tarball in the first place. So, yes it will work. No. It doesn't, because the .svn directory is not present and you get a Not under version control message. That's strange; how many people just responded with that worked? ___ --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] Voip providers and sip origination and termination?
Yair Hakak wrote: actually Rich, not to be picky or anything, but your first paragraph is backwards. There are some providers that allow you to originate calls to the US/World pstn network via their facilities, but do not provide any way for the US/World to call you from the pstn network. (eg, Origination only provider.) That's a termination only provider which allows you to terminate calls. otherwise, very informative.. Yup, I blew it. But, for the purposes of the OP, the point was made. ;) ___ --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] Polycom related question
John Marvin wrote: Rich Adamson wrote: If you look at the sample configs, you'll find: [EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple mailboxes in the sip.conf.samples for v1.2 stable. That is the only way that I know of to turn on the mwi for two different phones (eg, extensions). Is that what you're using and its not working? I think that is the opposite of what Kevin is trying to do. The above config is for one phone monitoring multiple voicemail boxes. Kevin wants multiple (two) phones monitoring the same mailbox, i.e. he is probably specifying the same mailbox within the config for each of the phones that will be monitoring that mailbox. I'm not sure why there would be any problems with that. Kevin, have you tried just having one phone at a time do the monitoring, to make sure there aren't any problems with the phone's config? When one misses a notification, is it always the same phone that misses it? It's interesting that the problem is intermittent, it would seem that if Asterisk doesn't support this that it would only notify one phone each time and that the results would be consistant. Phones don't monitor mailboxes. One needs to tell asterisk which phones are to be notified when a voicemail is left, and the sip statements above are the only ones that I'm aware of to accomplish that. On many phones, there is only one mwi function. If Kevin has one extn (eg, 111) on a phone set up with a mwi and then a second extn (eg, 222) on the same phone set up for mwi, one extn's mwi might turn the indicator on while the second extn will turn it right back off again. Since I don't recall Kevin saying what type of phone he's using, I can only guess that might be the problem. Its either that, or, my original comment above regarding the sip definitions. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using residential voip for business?
Christopher Corn wrote: I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled. is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because they have the unlimited offer for outgoing/incoming. Yes, that can happen. Read the fine print. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using residential voip for business?
Rushowr wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Christopher Corn wrote: thanks for the reply. why are residential lines cheaper than businesses? say for unlimited, it always costs more for residential. */Michael Graves [EMAIL PROTECTED]/* wrote: I'd just use a service that's being offered to business customers...like Nuvio's nPBX. While they don't support Asterisk directly some of their resellers will support using *. I've used it for about 6 months and its been very reliable. The only annoying thing is that they only support SIP connections. The rumour is that they may eventually offer an IAX2 based account for Asterisk users...but I've not yet heard if this is actually going to happen. FWIW, I ported my DIDs to Nuvio so that's where my incomming calls come from. I split my outgoing calls across Nuvio, Nufone Voxee. Michael --Original Message Text--- *From:* Christopher Corn *Date:* Sun, 10 Sep 2006 17:20:37 -0700 (PDT) i see. thanks for the info. */[EMAIL PROTECTED]/* wrote: Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off. -- Original message -- From: Christopher Corn [EMAIL PROTECTED] I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled. is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because they have the unlimited offer for outgoing/incoming. thx Typically a business offering costs more because the provider offers higher availability, reliability, call quality, etc... That's not true at all. I worked for a large telco for 20+ years (in all engineering disciplines), and the only reason business plans are more expensive then residential plans is that businesses generate more traffic. More traffic translates into more infrastructure costs (eg, central office equipment, trunks, etc). Businesses and homes generally use cable pairs (or fiber) out of the same cable, use the same central office line cards, etc. There is no difference in terms of availability, reliability or call quality. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using residential voip for business?
Michael Graves wrote: Not to answer a question with a questionbut why do so many businesses focus so intently on the cost of their voip service? If we presume that a business intends to stay in business, and that phone service is crucial to actually being in business, then I've never seen the wisdom of going to the absolute lowest bidder. Its because a lot of business managers don't understand telecom at all and the only thing they can relate to is dollars. Plus, most managers are beat about the head and shoulders about reducing costs, so their resulting focus is the lowest bidder. I once worked with a CFO in a very large telephone company. They wanted to redesign some of their internal financial reports to include items of primary interest. We had suggested removing Furniture and Office Fixtures from the report, and he objected. His comment was their are a lot of people working in the telephone company that didn't have a clue what the majority of the items on the report were, and we needed to leave the Furniture and Office Fixtures on them as that's the only thing those folks can relate to. (These were not technical reports either.) ___ --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 hangs up after 10-15 minutes whenSIPPhone is on mute
Several Linksys models have had a problem in the past allowing multiple devices on the inside lan to nat properly with something on the outside wan. Ordinarily a sip phone on the inside of the lan attempts to register with an external asterisk box, and the Linksys keeps track of source IP, source port, destination IP, and destination port. (That is part of every nat box.) But, on some Linksys models, they do not seem to track the source info, thus two sip phones appear exactly the same from an outside perspective. The issue can be seen in several forms including multiple sip phones, vpn clients, etc. Not sure exactly which models fall into the category, but I know from experience there have been multiple models over the years. You might also check to be sure your running the latest firmware on the Linksys. Mike wrote: That would be problematic. I am using a cheap Linksys router where my Polycom 501 is located and I see no such setting. It probably is hardcoded. Can I force the Polycom 501 to send empty RTP packet? (actually, I tried using comfort noise but I got an asterisk error message rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xx.xxx.xxx.xx Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using SIP to connect remote other VoIP server(Attn:Elpidio)
Crazy Boy wrote: Hi Elpidio, I am Chandra from India. I have a doubt. I am trying to solve my problem from many days. But, I couldn't able to solve this problem. I am using Asterisk with Redhat Enterprise Linux 4.2. The problem is 5060 port is blocked. After stop my firewall (service iptables stop) also, 5060 port is not opening. I checked with the below command: # nmap -p5060 192.168.91.22---This is my IP address and it is showing that port 5060 is closed. How can I enable and open this 5060 port? Really, I am breaking my head with this problem. SIP is not working because of this problem. Please tell me a solution. Looking forward to your reply. Thank you. The quickest way to determine whether an application is listening on a port is to simply do a 'netstat -an' from the linux command line. You should see something like this: udp0 0 0.0.0.0:50600.0.0.0:* If you don't see that, then asterisk is not opening the port. From an asterisk command line, do 'show modules like sip' and you should see something like this: Module Description Use Count chan_sip.soSession Initiation Protocol (SIP)0 If you don't see that, then asterisk is not loading the chan_sip.so module for some reason. Look in /etc/asterisk/modules.conf and make sure there is NOT an entry in that file that looks something like this: noload = chan_sip.so If that entry is not there, then you either have a problem with the configuration of the file /etc/asterisk/sip.conf, or, some other problem that is causing asterisk to not load chan_sip.so. If you are sure the sip.conf is absolutely correct and error free, then stop asterisk, and start it from the linux command line with 'asterisk -c'. There should be some indication why chan_sip.so is not be loaded, etc. ___ --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] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls
Dan Serban wrote: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be xfer'ed via the soft button on the phone itself, it seems that if you hit the button twice in quick succession, there is no problem (effectively a blind transfer), if then I try to tell the other extension that Joe is calling to sell you a fridge and hit xfer, the calling party cannot hear what that person at the extension is saying. Sometimes the tables are fully turned, the caller can hear, but the operator can't hear a thing. One thing's for sure, if you hit the button quickly (blind transfer) it works no problem at all. This is what I see asterisk saying when I transfer the call unsuccessfully. == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' I've looked at the macro with a fine tooth comb, I cannot see any problems with it whatsoever, (though that doesn't mean that my ignorance isn't getting in the way). I found some mention on the digium mantis bug tracker, here's the link: http://bugs.digium.com/view.php?id=7421 Before I try and patch the source (which I'm hesitant to do since I run the debian packages), is there another solution or maybe an unidentified issue that I haven't been able to decipher? I had somewhat the same issue today, but in my case any attempt to do the transfer resulted in a busy signal. Removing call-limit=1 from the sip definitions corrected the problem. Since you are getting past the point of dialing the extension that you are transferring the call to, the problem sounds more like either phone configuration issues or something in the network blocking data flows in one or the other direction. Are all the phones and asterisk on the same network with no firewalls, nat, etc? As far as hitting the transfer button twice, that doesn't work here. The first press for the transfer is the transfer function, but the second press results in -dnd (since the softkeys changed right after the first press). The spa941/942's here are almost 100% default config's running the latest firmware (as of a couple of days ago). I'm running v1.2.10 also, however the source code was installed via SVN checkout and I've done several 'make update' to pull the most current fixes. ___ --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] ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server
Marco Mouta wrote: Hi all, Do you think it could be an affordable solution using a two fxs ATA device to connect an old legacy pbx (with few users) with a main asterisk server. phonesanalogueSmallOfficeLegacyPBxATA-2FXS-SIP--MainOffice AsteriskServer This way also I would use ATA device as a Trunk without requiring an Asterisk server on every smalloffice and no need to buy many ATAs neither VoiP hardphones. Is this affordable or i'm missing already basic functions required for a production system? One item you will need to research and tends to create problems for people doing this is line supervision. In other words, disconnect supervision, answer supervision, etc, are often times not provided by legacy pbx's, and therefore the ATA may not recognize hangups, etc. ___ --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] ONE WAY VOICE ONLY IN ASTERISK
Elpidio Ramos wrote: This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 1 thru 3) for both interfaces. I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop. This is my sip.conf file: [general] context=ramosoft allowguest=no realm=ramosoft.com bindaddr=0.0.0.0 bindport=5060 srvlookup=yes pedantic=yes tos=184 tos=lowdelay maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=ilbc allow=gsm musicclass=default language=es relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 useragent=RamoSoftPBX regcontext=ramosoft localnet=10.10.10.0/255.255.255.0 rtcachefriends=yes [authentication] [311] type=friend regexten=311 username=311 secret=311 callerid=Elpidio Ramos 311 host=dynamic nat=yes canreinvite=no Is there anything I am missing here to get two way voice? Thank you in advance all If you have two working nic's, then when the soft phone is on the inside of the network, it should register with the IP address of the inside nic. When the soft phone is on the outside (eg Internet), then it should be registering with the IP address of the outside nic. Any other combination is going to give you problems and particularly if you are using a firewall. The problems will be associated with basic layer-3 stuff and nating. ___ --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] UK (BT) Problem with TDM 400P
Mark Muffett wrote: I'm trying to get my TDM400P to work with a BT POT line. I've done everything I can think of to get the uk settings right (in zapata.conf, zaptel.conf and options for the wctdm driver) - and they all look right (ie uk like) and look like they are working when I try diagnostics with lspci, dsmsg, asterisk -c etc - but when I connect the FXO unit to the BT line it just makes it (the BT line) go permanently engaged. I'm tearing my hair out and about to chuck it all in the bin, but before I do, has anyone ever managed to get a TDM400P to work with a BT line and did they have any of these issues? Thanks for any help Bad fxo module? Call digium support and let them help diagnose the problem. Also, the E/F revision is rather old; current is more like rev J. ___ --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] Polycoms, Attended Transfer and Canreinvite = yes
The svn branch-1.2 is very stable, probably more stable then the rpms and other distro's out there, as fixes are applied when problems are identified and corrected. Sometime later, the svn branch-1.2 is used to create packages. Kevin Smith wrote: Well personally I am just glad I wasn't the only one seeing the problem. As much as I don't like the place 100% of the blame on something unless I fully know what is going on, in this case Asterisk, but I couldn't see any solution but a bug. Personally I wouldn't mind testing out the branch, but I know my boss, isn't so trusting. How stable are the SVN branches, at least in terms of justification for taking the system down to install it? Or is there an easier way to test? Thanks, Kevin Kevin P. Fleming wrote: - Richard Scobie [EMAIL PROTECTED] wrote: Dave Fullerton wrote: I just verified it here as well. Running Asterisk 1.2.11 and two polycom I'll throw in a me too here, with the addition that it also occurs with canreinvite=no. There were multiple problems in this area, introduced since Asterisk 1.2.9 was released. We believe that with today's commits in SVN branch-1.2 they are cured, so it would help us greatly if could download SVN branch-1.2 and try it out on your system to see if it solves your issue. I apologize for how this crept into the code base... it should not have happened, and we are taking steps to ensure that future changes in the release branch don't cause regressions like this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura SPA3000
I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. ___ --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] Sipura SPA3000
Steve Kennedy wrote: On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. On power failure, yes. On ethernet cable disconnect, yes. But, when asterisk simply does not respond (for any reason), no. The last part is the difficult part. There isn't any logic in the spa3k that would essentially ping the asterisk service to see if it responds, and then do some alternate action if it does not respond. (One easy way to confirm that is to look around the spa3k config and see if you can find anything that relates to sip failure fail-over, timing entries associated with detecting a sip failure (lack of response), etc.) ___ --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] Sipura SPA3000
If pstn call ring thru line 1 is enabled, all incoming pstn calls will ring through to the fxs port (and not to asterisk). The OP was looking for a auto fail over function that essentially would be pstn call ring thru line 1 on sip failure. That doesn't exist. Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve ___ --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] Sipura SPA3000
That option addresses what to do with the fxs (line 1) when the registration fails as opposed to what does the fxo (pstn line) does when registration fails. Bob Chiodini wrote: Rich, After reading a little more, how about the Line 1 VoIP Fallback to PSTN (section 4.9)? It looks like this is invoked when the Ethernet link is down or registration fails. I don't have a SPA3000 up at the moment to look at what's required. Bob... On Fri, 2006-09-01 at 11:45 -0500, Rich Adamson wrote: If pstn call ring thru line 1 is enabled, all incoming pstn calls will ring through to the fxs port (and not to asterisk). The OP was looking for a auto fail over function that essentially would be pstn call ring thru line 1 on sip failure. That doesn't exist. Bob Chiodini wrote: Probably the PSTN Call Ring Thru Line 1 feature. Section 4.11 in: http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf#search=%22spa3000%20manual%22 By default, if my asterisk went down after the SPA3000 was already registered, the in-bound PSTN call was lost. I probably did not wait long enough and I did not have PSTN Call Ring Thru Line 1 enabled. Bob... On Fri, 2006-09-01 at 10:32 -0400, Kevin Collins wrote: My 3000 does this natively without config. Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Friday, September 01, 2006 10:03 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sipura SPA3000 On Fri, Sep 01, 2006 at 08:11:50AM -0500, Rich Adamson wrote: I have a question on configuration of SPA3000 with asterisk. 1. I want all incoming calls are redirected from SPA3000 to my asterisk server. 2. Asterisk then should direct this call to my SIP phones (including Sipura) 3. In case asterisk server is down I want that call be directed straight to the handset connected to the Sipura Is this configuration possible? The spa3000 does not have logic in it to support #3. I thought the SPA3K could do this, i.e. on power failure or non-ability to connect to server, connect FXS to FXO. Steve ___ --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] oddity with TDM400P / Asterisk setup
Ted Wallingford wrote: Hi List, I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything. The * version running is 1.2.7.1. All of the ports on the switch with voice devices, including the server, have a service class of 5, while non-voice devices are connected to other ports that have a service class of best effort. The problem, which began this morning, is very elusive. Calls-in-progress from zap-to-sip or sip-to-zap or sip-to-Asterisk will drop at odd times during the call, anywhere from 2 minutes to 15 minutes into the call. At the same time the call drops, my SSH session to the server will hang. After 10 to 15 seconds, the output and input from ssh session appears on my terminal and I am able to resume working in the shell. Zap-to-Asterisk doens't seem to cause the problem. Only when I dial through to a SIP device does it seem to hang. Top reveals nothing out the ordinary, utilization wise, the disk has plenty of free space, and the arp cache doesn't ever indicate a duplicate IP address with the server's NIC, which I thought might have been the problem. I also attempted to move the server to another port on the switch. No improvement. Anybody have a problem like this? Have not seen anything close to that problem. You might check the linksys switch to see if it has Spanning Tree turned on. Spanning Tree (depending on vendor code) will disable a port from forwarding traffic for about 10 to 15 seconds as a means of detecting layer two loops. If it is turned on, turn it off and test again. Also, you should be able to set up a series of pings from different sources to determine exactly which component in the infrastructure is failing. ___ --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] HP ProLiant and Digium 24xxp
Kevin P. Fleming wrote: Robert Roach wrote: I have a customer request to deploy an HP rack server (ProLiant DL series) as the base system for an Asterisk install. They also want to use the Digium 24xxp card. I have heard that the Digium card is oversized and does not fit in a normal size chassis. Does anyone know if it will fit in the ProLiant chassis, or have a recommendation on another HP box to use? This is incorrect. Nobody (including Digium) makes 'oversized' PCI cards, because there are no chassis in the world they would ever fit in. However, it is true that the TDM2400P is the maximum possible size of a PCI card, both full-length and full-height. In addition, it requires a standard hard-drive power connector (Molex) to supply 12V power if any FXS modules are used, which are often hard (or impossible) to find in a 1U or 2U rack-mount server. There are some available, though, and shortly Digium will have an external power solution available for the TDM400P and TDM2400P cards. The issue is not the TDM2400 is over sized, but rather some PC hardware vendors assuming no one ever uses full sized cards anymore. Lots of systems have crowded fixed drive bays and other stuff into their cases that preclude using full sized cards (from any source, not just the TDM2400). And as Kevin just mentioned, they also assumed there will never be a need for a Molex power connector in the pci bus area of the box. ___ --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] iax vs. sip?
We've been using iax with teliax.com for a couple of years, and it seems the quality of calls varies with time. Sometimes it is good and next time its not so good. There has been changes occurring to iax and the jitterbuffer stuff over the last two years, and I'm reasonably certain that some poor quality is related to differences between teliax.com's implementation (eg, s/w versions) and ours. I've not bother to try sip since our asterisk implementation is truly both a production box for our small office, and a test box for various version testing, etc. We used iax for more than a year and moved to sip about 6 months ago. The quality from termination providers seems much better now with sip. Tom At 09:38 PM 8/30/2006, you wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIP might be better to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. ___ --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] Tracing audio problems
Avi Miller wrote: Avi Miller wrote: Does anyone have any suggestions on where to look next? My users are getting increasingly annoyed and I'm quickly running out of ideas. Replying to myself to note that this is now happening on outbound calls via ISDN, i.e. calls that don't use IAX2 or the inter-office network. It also happens on inbound calls. Is this a new installation, or, were the boxes working okay for a while and they just now started having problems? Based on your comments that poor audio seems to only be occurring on the isdn calls, it would almost sound like the T1/E1 card's clock sync parameter wasn't right. Check /etc/zaptel.conf for: span=1,1,0,esf,b8zs where the second 1 indicates this particular span is to be used for syncing the on-board clock. (I don't use sangoma's T1/E1 card, so not sure if /etc/zaptel.conf it the right place.) Are the poor audio calls always associated with one site (head office)? What does 'zap show status' indicate at those sites that have bad audio? Do you have iax links to these sites as well, and if so, are you having the same audio problem with them? What type of phones are you using to initiate the calls with bad audio (sip phones or what)? ___ --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] lost packets when bridging zap and iax
Simone Cittadini wrote: We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller -- ( zap - iax ) --- ( iax - whatever ) -- called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats on client machine shows more and more dropped packets on the local side if we use sip as the entering point for the calls all works well : caller -- ( sip - iax ) --- ( iax - whatever ) -- called client server seems something in the bridging between zap and iax screws up, but I don't know if it's a bug or a misconfiguration, my conf files follows, someone has similar experiences to share ? /etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=xxx.xx.xx.xxx disallow=all allow=alaw jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes language=it notransfer=yes /etc/asterisk# cat sip.conf [general] context=invalid bindport=5060 bindaddr=xxx.xx.xx.xxx srvlookup=no disallow=all allow=alaw progressinband=no canreinvite=no language=it [authentication] [some-ip] type=friend context=ip host=some-ip /etc/asterisk# cat zapata.conf [channels] language=it context=default switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe immediate=no callerid=asreceived usecallingpres=yes echocancel=yes echocancelwhenbridged=no ;echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 group = 1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 /etc/asterisk# cat /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 span=2,1,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,1,0,ccs,hdb3 bchan=63-77 dchan=78 bchan=79-93 span=4,1,0,ccs,hdb3 bchan=94-108 dchan=109 bchan=110-124 loadzone=it defaultzone=it Only one of the above four span entries should have a 1 as the second digit. That second digit is telling the digium card which span to sync its on-board clock to. Pick the span that goes to a central office and specify it as 1 and all other spans should be either 0 or increasing numerical digits (eg, 2,3,4). If none of the spans go to a central office, its still a problem. You'll have to reload the drivers for the change to take effect. ___ --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] Missing number 2 in advanced options of VM
Doug Lytle wrote: Stefan-Michael. Guenther (in-put GbR) wrote: Why does Asterisk strip all digits except 4498 and why doesn't _X. match That I can't answer, I've never used the option. My VM works just fine by sending the callback through the same context as what your sip phones use. ___ --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] Can anyone recommend a large button sip phone for the elderley.
Chuck Bunn wrote: Hi, Can anyone recommend a large button/type sip phone (VOIP) that an older person could use. I have a client that needs to have large button phones for elderly residents in her facility. How about the old Grandstream BT100? Large buttons, requires a firm press (no nervousness), no fancy features. ___ --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: RE : [asterisk-users] Problem with a TDM400P
Seems to me that someone posted something about unusual analog connections in the UK that required a jumper wire (or something like that) on the pstn analog connection to the fxo port jack. (I'm in the US, so don't have a clue what I'm taking about.) Might be worth doing a little more google searching on that thought. Mark Muffett wrote: Changed it round now, FXS on 1 2, FXO on 3, but still the same problem. Any ideas for diagnostics? On 28/08/06, Mark Muffett [EMAIL PROTECTED] wrote: Never heard that - but come to think about it, all the configs I've seen are like that. I'll give it a try and let you know. Thanks Mark On 28/08/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello Mark and the list, What about if you change the order of the modules, starting with FXS first and finishing with FXO on the TDM400P slots ? I remember to have read something like always start with FXS if FXS and FXO modules are present on the board... Feedback please. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Mark Muffett Envoyé : lundi 28 août 2006 19:49 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Problem with a TDM400P I'm setting up my first (and very simple) Asterisk PBX and running into problems with the FXO module I have on a TDM400P - I'm trying to connect to a standard UK, BT, POT. The problem is that when I plug the FXO module into a functioning BT line, it seems to make the line become engaged - ie if I try to call it from another number I just get the engaged tone. This happens whether or not asterisk is running and even whether or not the zaptel modules are loaded. The TDM400P card seems to be ok - I get the expected line: 04:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface when I type lspci. My FXO module is in position 1, with FXS modules in positions 2 and 3. The FXS modules seem to work ok with my config files (I get a dialing tone if I connect a phone to them). My zaptel.conf file is simply: fxsks=1 fxoks=2,3 loadzone=uk defaultzone=uk and my zapata.conf is (at the moment): [channels] ; context=test usecallerid=yes hidecallerid=no immediate=no signalling=fxo_ks echocancel=yes group=1 channel=2 channel=3 signalling=fxs_ks echocancel=yes busydetect=yes answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=yes group=2 channel=1 (I've tried getting rid of busydetect, answer/hanguponpolarityswitch, and callprogress individually and all together). Could I have a hardware fault? - if so any ideas what tests to run? Or is there something else I need to configure. Thanks for any help. Mark ___ --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] zap channel media volume
If one would visit with knowledgeable transmission engineers that work full time in the telephone industry, one would find telephony standards that govern exact transmission levels at each point throughout a country's telephone network (including the long distance facilities, pbx trunk loss, CO switch loss, etc). The only variable in those standards are the end user loops, which varies due to the length of the loop and other mostly uncontrollable and/or variable factors. The individual telephone companies oftentimes have internal transmission standards that govern what is or is not acceptable in terms of end user pstn loops. Practically all US telcos of any size force their installers to measure the transmission loss for every new installation, and oftentimes on any repair call. Asterisk's pc-based analog I/O cards totally ignores those standards. So, an automatic gain control would be nice but it would really be a work around for other root-cause / design problems. In testing various analog pstn I/O cards, I've found the sangoma A200D card (with hardware echo canceler) to be the best pstn analog interface on the market that address both the echo and transmission level issues for the longer higher-loss pstn loops. Transmission levels are still a little bit low but very usable. JD Austin wrote: I've been struggling with this issue for over a year. I wish there were some kind of automatic gain control built in to set the rx/tx gain on the fly based on the volume of the two channels. Probably not realistic though. Is there other hardware other than digium's that better deals with this issue? Rich Adamson wrote: The root cause of the low volume problem is the result of software echo cancellation software, and its need to insert a noticeable loss. If I recall correctly, the wctdm.c driver has a statically defined loss value of something like -6 db that is loaded into the TDM400 chipset at driver load time. Ordinarily, that loss is not all that noticeable. But, if your pstn line is rather lengthy (greater then about 5db worth of loss), the two loss values become very noticeable and marginal to users. There is no known fix or workaround. The low audio becomes even worse when a pstn caller leaves a voicemail and the user calls in via the pstn to retrieve his voicemail. The voicemail gain setting was intended to be sort of a workaround, but its marginal at best. JD Austin wrote: I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? ___ --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: GSM gateway and FXO ATA
Martin Joseph wrote: On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said: Hi list! I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over Grandstream HT488 ATA. snip Personally I found the FXO port on the HT-488 to unworkable except as a backup for power outages. I found several problems with it. 1) serious echo issues (I have a long loop). 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. 3) The device hangs and reboots itself occasionally. This is all just an FYI. Marty PS I did test with the latest HT-488 firmware and all issues were still present. I'd agree with the above 1000%. It should be taken off the market. ___ --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] Trunk with multiple IPs?
I don't believe that addresses the OP's original post since he was talking about limiting incoming calls from specific IP addresses. You might want to validate how secure your definitions are considering the type=friend approach. Lists @ EMS wrote: Hi, I've only just now seen this post. This is how we have setup. In sip.conf [xxx.xxx.xx1] host = xxx.xxx.xx1 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx2] host = xxx.xxx.xx2 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx3] host = xxx.xxx.xx3 type = friend insecure = very context = your-context canreinvite=no [xxx.xxx.xx4] host = xxx.xxx.xx4 type = friend insecure = very context = your-context canreinvite=no Hope this helps. Paulo I wish I could offer some direct help on whether or not your method with a comma separated list would work, but I can't. However, you could always create a few entries using different formats and then run some tests against them Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --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] DNS
Bill Gibbs wrote: Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to start working while I fixed the primary. Is this behavior normal or am I missing a step? All hosts, etc are identified by IP. Ver 1.2.10 Most people don't think much about dns, but if your primary dns server responded with anything (including a simple I don't know response), the secondary dns server will not be attempted. So, depending upon exactly what was wrong with your primary, your stated result can be very normal. Regarding asterisk stop responding when no dns server is present, that's been discussed many many times on this list, the latest as of earlier this week. Asterisk code does have a problem, and I'd be reasonably certain part of the problem is the OS underlying dns resolver operates in a blocking mode. In the past, one of the suggested workarounds was to implement a dns caching-only server on the asterisk box. I've not done that and I don't recall hearing anyone's actual experience after doing it. Another suggested workaround is to use IP addresses only in your configs (which is what I've been doing for three years). But, you'll need to make sure nothing in the configs gets interpreted as a dns name. ___ --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] zap channel media volume
The root cause of the low volume problem is the result of software echo cancellation software, and its need to insert a noticeable loss. If I recall correctly, the wctdm.c driver has a statically defined loss value of something like -6 db that is loaded into the TDM400 chipset at driver load time. Ordinarily, that loss is not all that noticeable. But, if your pstn line is rather lengthy (greater then about 5db worth of loss), the two loss values become very noticeable and marginal to users. There is no known fix or workaround. The low audio becomes even worse when a pstn caller leaves a voicemail and the user calls in via the pstn to retrieve his voicemail. The voicemail gain setting was intended to be sort of a workaround, but its marginal at best. JD Austin wrote: I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? ___ --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] DNS
Ola Lidholm wrote: On 25 aug 2006, at 20.18, Bill Gibbs wrote: Asterisk server is setup in /etc/resolv.conf to query my primary and backup NS. Had an issue with my primary NS and asterisk refused to complete any calls or forward inbound calls to extensions. I had to manually switch it to look at the backup NS first then reboot for it to start working while I fixed the primary. Is this behavior normal or am I missing a step? All hosts, etc are identified by IP. I have had similar issues. To sort of resolve this I had to install a local name-server on the machine that contains the addresses asterisk tries to resolve (changing to using IP-addresses did not fix the issue for me either). I would prefer an option in asterisk that tells it to not resolv more than once on each address. That won't fix the problem. If that's all you needed, then change your resolver to use /etc/hosts and statically define each item. However, that totally defeats the dynamic purpose of dns. If you configure the dns server (on each asterisk box) to be a caching only server, then it will do the normal dns lookup and cache that translation one time. Asterisk is generally happy with that. However, if the owner of the dns name that you're looking up sets an unreasonable time-to-live for that name, the caching server isn't going to help much on a flaky network. ___ --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] Cisco Router QOS and IAX2
The majority of the sample qos policies seem to be based on either five or seven qos queues, and most folks don't need all of that. What I've shown as a sample only has three queues; one for voip, one for my outbound web traffic, and the default queue that everything else falls into. You can actually remove the sections relating to web traffic if you don't have a production web server contending for outbound traffic, making it a two-queue policy. R. Bruce Reeves wrote: Thank you so much. After fighting with a large/extensive QOS policy from Cisco's SDM tool, I used your sample and tweaked it for my needs and everything started working fine. Bruce On 8/23/06, *Rich Adamson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I like using the TOS header bits personally as lots of other protocols (eg, dns) will eventually match on udp/4569. For the TOS bits v1.2.10, use tos=lowdelay in iax.conf and on the cisco use an access list to match on the tos bits. Something like: access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay = same as tos=lowdelay access-list 103 permit ip any any tos 12 For the TOS bits svn truck, the tos= settings have changed in asterisk. Look in the supplied documentation (eg, readme's, sample configs) for exactly what is allowed in terms of DiffServ (new term for TOS basically). You'll find examples that support the above access list item dscp cs3 and dscp ef. If you're not all that experienced on cisco qos, then the following is an example of a working config that you should be able to translate into your router config one way or another. class-map match-all voice-rtp match access-group 103 class-map match-all www-traffic match access-group 105 ! policy-map voice-policy class voice-rtp priority percent 40 class www-traffic bandwidth percent 30 class class-default fair-queue ! interface Dialer0 bandwidth 555 snip, my specific interface config statements service-policy output voice-policy ! access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www any The above config provides low-latency priority to voice-rtp, then provides an additional qos piece to ensure www-traffic is given bandwidth before all of the class-default traffic. In other words, voice-rtp traffic will always get 40% of the bandwidth (eg, 40% of bandwidth=555 above) if voice traffic is present. If voice traffic isn't present, that bandwidth can be used by other qos sections or by the default class. Same with www-traffic after the router deals with voice-rtp traffic. The default class always gets what bandwidth is left over (or all bandwidth if there is no voice-rtp or www-traffic). To troubleshoot the above, do a show access-list 103 from the CLI (on the router) and watch for matching packets in each access list line. Once you've structured the access list to truly match asterisk traffic, then do a show policy-map interface dialer0 to display how the overall qos structure is functioning. Note that cisco didn't get real serious about IOS qos until v12.2 of their IOS code. In v12.2 (and later versions of IOS) there has been a significant amount of work to bring all of their products into industry standard implementations / conformance / expectations. If you want to get real serious with cisco's qos stuff, purchase the book End-to-end QoS Network Design and read the 700+ pages devoted to the subject. It is an excellent book with lots of examples, etc. The book (and actual practice) suggests IOS v12.3 has more QoS funtionality then v12.2 , and v12.4 has more then v12.3. (The authors of the book back that statement up 100% as well, and they are cisco employees.) In the above config, the bandwidth=555 statement is very important. It should represent the actual outgoing bandwidth for whatever interface you are using and not the theoretical max that someone said you should get. Also note that for relatively slow speed interfaces (eg, most dsl's), the outgoing bandwidth is rather slow. If you calculate how much time is consumed sending a non-voice 1500-byte packet, the time is likely to be more then the 20 millisecond interval
Re: [asterisk-users] Modems dialing over sangoma a104d
Sean Cook wrote: I have a sangoma 104d that is our main pbx now( legacy system died ). I have replaced every phone in the building and things are going very well. We have fax working well and calls are routing properly... All is well... Except for our support modems... we have support people that dial out with modems across our PRI's. These modems are attached to an Adtran 750 with 24 FXS's. I have disabled echo cancelation on the T1 that is connected to the Adtran but negotiation is still really rough. I am bridging across the same card and it isn't doing very well... has anyone done this with reasonably successful results? I am not looking for 56K I am looking for around 9600 to 14.4.. Can we assume that you've got the correct timing parameters set on the 104d? (eg, are you sync'ing your 104d from the telco?) If not, get that corrected first as it makes a major difference with modem calls. The echo cancellation disabling should be automatic I believe, so would not expect turning it on/off to have much of an impact. I've done this with the sangoma's analog card (a200d) and modem calls work very well. I didn't actually check the speed, but felt like 14.4 or better. One of the stated test suites (by sangoma employees) is to validate all hardware and driver designs by testing with modems since that really is one of the most critical non-test-equipment tests that can be done. So, I've got to believe it works; just need to identify the missing link. ___ --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] Trunk with multiple IPs?
Benjamin Lawetz wrote: Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? Here's an iax.conf example of what I'm using: [teliax] context=teliax-incoming type=user auth=md5 secret=mysecret jitterbuffer=yes disallow=all allow=gsm deny=0.0.0.0/0.0.0.0 permit=207.174.202.0/255.255.255.0 The last two statements essentially restrict incoming calls from teliax to one of their class-c networks (regardless of how many servers or IP's they have). Note that on incoming calls the host= line is not used. If you're really asking how to do that for outgoing calls, you'll probably have to do it through three/four sections (type=peer) and deal with those sections in your dialplan. As a side note, there are a large percentage of * implementors that don't understand the search terms when an incoming call is being negotiated (eg, is host= used, is secret= used). Without that understanding, calls likely come into different sections then what the implementor actually expected. The deny permit statements are very useful to tighten down security for each incoming context. ___ --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] Calls over VPN
Joseph wrote: Is anybody making calls over VPN? If so what is the penalty as encryption is involved. I was planning to use VPN to register Sipura units to my local asterisk this way I don't have to deal with NAT issues. vpn's work just fine as long as the vpn end-points have enough horsepower to encrypt/decrypt packets without delay. As one example only, some of the older cisco routers didn't have enough horsepower to sustain any significant amounts of vpn traffic without installing an optional vpn hardware card. I'd have to guess this might also be true with some inexpensive boxes like Linksys as well. ___ --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] VM - advanced options?
running v1.2.10 svn checkout... When I listen to the VM options, it says 'press 3 for advanced options', but after pressing '3', there is nothing there with the exception of pressing '*' to return to the main menu. Have I missed a config option, sound file, or is the advanced option not totally implemented as yet? R. ___ --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] Cisco Router QOS and IAX2
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I like using the TOS header bits personally as lots of other protocols (eg, dns) will eventually match on udp/4569. For the TOS bits v1.2.10, use tos=lowdelay in iax.conf and on the cisco use an access list to match on the tos bits. Something like: access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay = same as tos=lowdelay access-list 103 permit ip any any tos 12 For the TOS bits svn truck, the tos= settings have changed in asterisk. Look in the supplied documentation (eg, readme's, sample configs) for exactly what is allowed in terms of DiffServ (new term for TOS basically). You'll find examples that support the above access list item dscp cs3 and dscp ef. If you're not all that experienced on cisco qos, then the following is an example of a working config that you should be able to translate into your router config one way or another. class-map match-all voice-rtp match access-group 103 class-map match-all www-traffic match access-group 105 ! policy-map voice-policy class voice-rtp priority percent 40 class www-traffic bandwidth percent 30 class class-default fair-queue ! interface Dialer0 bandwidth 555 snip, my specific interface config statements service-policy output voice-policy ! access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www any The above config provides low-latency priority to voice-rtp, then provides an additional qos piece to ensure www-traffic is given bandwidth before all of the class-default traffic. In other words, voice-rtp traffic will always get 40% of the bandwidth (eg, 40% of bandwidth=555 above) if voice traffic is present. If voice traffic isn't present, that bandwidth can be used by other qos sections or by the default class. Same with www-traffic after the router deals with voice-rtp traffic. The default class always gets what bandwidth is left over (or all bandwidth if there is no voice-rtp or www-traffic). To troubleshoot the above, do a show access-list 103 from the CLI (on the router) and watch for matching packets in each access list line. Once you've structured the access list to truly match asterisk traffic, then do a show policy-map interface dialer0 to display how the overall qos structure is functioning. Note that cisco didn't get real serious about IOS qos until v12.2 of their IOS code. In v12.2 (and later versions of IOS) there has been a significant amount of work to bring all of their products into industry standard implementations / conformance / expectations. If you want to get real serious with cisco's qos stuff, purchase the book End-to-end QoS Network Design and read the 700+ pages devoted to the subject. It is an excellent book with lots of examples, etc. The book (and actual practice) suggests IOS v12.3 has more QoS funtionality then v12.2, and v12.4 has more then v12.3. (The authors of the book back that statement up 100% as well, and they are cisco employees.) In the above config, the bandwidth=555 statement is very important. It should represent the actual outgoing bandwidth for whatever interface you are using and not the theoretical max that someone said you should get. Also note that for relatively slow speed interfaces (eg, most dsl's), the outgoing bandwidth is rather slow. If you calculate how much time is consumed sending a non-voice 1500-byte packet, the time is likely to be more then the 20 millisecond interval for sip/iax traffic. If that is your case, then you may need to forcibly reduce the MTU size of packets originating from other non-voice workstations/servers. The later cisco IOS versions have a parameter to do that if you can't do it via the workstation/server configuration parameters. If memory serves correctly, that parameter appeared around v12.4 of their IOS. One last item... all of the above deals only with outgoing traffic. You would need to talk to your ISP about QoS for your incoming traffic, and most of the local ISP's don't have a clue. Increasingly, some of the larger backbone isp's are beginning to understand QoS and some have actually implemented something. However, those isp's are heading towards providing QoS as a value-add chargeable service (as in MPLS, etc). R. ___ --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] Trunk with multiple IPs?
I'm thinking I used deny and permit statements on broadvoice.com way back when, and the configs/sip.conf.sample suggests its still valid for v1.2.10 code. You might take another look at that for sip. Benjamin Lawetz wrote: Agreed that with a other IAX and SIP that have registration information and secrets that works. The problem is when you have a provider that just sends you a SIP call and the only way to identify it is by IP address. In those cases (if I understand correctly) we need a host line don't we? (Or at least I remember when I was testing a while back that it wouldn't work with deny and permit) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: August 23, 2006 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trunk with multiple IPs? Benjamin Lawetz wrote: Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? Here's an iax.conf example of what I'm using: [teliax] context=teliax-incoming type=user auth=md5 secret=mysecret jitterbuffer=yes disallow=all allow=gsm deny=0.0.0.0/0.0.0.0 permit=207.174.202.0/255.255.255.0 The last two statements essentially restrict incoming calls from teliax to one of their class-c networks (regardless of how many servers or IP's they have). Note that on incoming calls the host= line is not used. If you're really asking how to do that for outgoing calls, you'll probably have to do it through three/four sections (type=peer) and deal with those sections in your dialplan. As a side note, there are a large percentage of * implementors that don't understand the search terms when an incoming call is being negotiated (eg, is host= used, is secret= used). Without that understanding, calls likely come into different sections then what the implementor actually expected. The deny permit statements are very useful to tighten down security for each incoming context. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk + sangoma a102 to simulate telco PRI: is possible?
Giorgio Incantalupo wrote: Hi, I have an asterisk box with a sangoma a102 (two PRI ports). Is is possible to connect port A to port B in order to use port B as a simulation of a telco PRI line? If yes, is there a special cable needed? How can I configure the card and zaptel.conf? Yes. You'll need a T1 crossover cable to do it. Google for which pins to swap. Configure one port as pri_net (acts as a central office switch) and the other port as pri_cpe (acts as a pbx). See the sample configs for other parameters (including /etc/zaptel.conf timing parameters). Your zapata.conf entries will look something like these: resetinterval=never ; gets rid of the many restart messages context=pri-in signalling=pri_net switchtype=national pridialplan=unknown channel=1-23 context=pri-out switchtype=national signalling=pri_cpe pridailplan=unknown group=7 channel=25-47 And, /etc/zaptel.conf something like this: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 ___ --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: Zaptel install - Fedora Core 5
Tzafrir Cohen wrote: On Tue, Aug 22, 2006 at 08:42:36AM -0500, Rich Adamson wrote: Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I did yum update last week and here is my current kernel: I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Hi Anto! The thing is that I can't rely on yum update for asterisk installation. I would like something that will work like this: I install FC5 from CD/DVD, install RPM's that I need from my ftp server or from CD, install zaptel, libpri, asterisk... So, I need to download rpm's that will allow me to install zaptel/libpri/asterisk without using yum update (I need to make all installations the same). Why bother with the rpm's? Because you have some other programs on your system other than Asterisk. And because you want a reproducable build. Guess that depends a lot on personal objectives, styles, and whether asterisk code has been modified locally. Once the reproducable build is operational and one has to maintain the code, reproducable builds sort of go out the window (eg, customer/system A has a problem, but not customer B through Z). Using the Branch SVN checkout approach always provides the most up to date code as opposed to replicating buggy stuff via rpms. ___ --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] LOUD MP3 Hold Music
David Freeman wrote: I have the opposite problem. I can hardly hear the hold music at all. On 8/22/06, *Dennis P. Clark* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How do you lower the volume of MP3 hold music? I'm certainly not an expert on MOH, but I don't believe there are any volume control knobs to be tweaked in asterisk itself. You might want to take a look at the configs/musiconhold.conf.sample file as there were some parameters that impacted high/med/quiet modes, but I'm thinking they only applied to the old mpg123 music app. Could easily be wrong. Best guess... probably have to use something like sox to change the volume of the mp3 (or whatever) file itself. ___ --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] Strange SIP response
Diego Andres Asenjo G. wrote: Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. Check the sip device config and make sure Do Not Disturb (DND), Call Forwarding, etc, have not be set. ___ --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] No retry after DNS failure
John Marvin wrote: Today I had a brief power outage which caused the Asterisk server and DSL modem to reboot. The Asterisk server came up before the internet connection was working, so it failed when try to look up some of the hosts for my outbound voip providers in sip.conf. Asterisk never recovered from that, i.e. it never retried so those providers were unavailable. The only provider that was still available was one that I had entered the IP address for, rather than the host name. Have any of you run into this issue, and if so, how have you solved it? It seems that since Asterisk periodically tries to reregister it should also retry the DNS lookup at the same time, rather than never trying again if the lookup fails. This would indicate that Asterisk would also fail if the voip provider changed the IP address of its server because Asterisk would never see the new IP address. Here are some workarounds I thought of, but none of them are particularly good: 1) Get a UPS so my machines won't reboot when the power fails. This actually might not solve anything, because I'm connected to a remote DSLAM in my neighborhood that I believe does not have backup power, so it won't work when the power is out. But perhaps Asterisk is more robust after it has booted (I'll have to test this). 2) Change all host names in sip.conf to IP addresses. This is kind of ugly and also will break when a voip provider changes their IP address. There is a reason for DNS! 3) Have a cron job send asterisk periodic sip reload commands. 4) Delay the start of asterisk until the internet connection has come up. This could cause me to be without any phones if there is any delay or failure in bringing up the network (I also have zap channels and PSTN lines). 5) A hybrid of ideas 3 and 4 above: Have a startup script that waits for the internet connection to come up, and then sends a sip reload command to Asterisk. Any other ideas? If memory serves correctly, most of the above has been raised as issues in the past and the suggested work around has been to run a dns caching server on the asterisk box. FWIW, I always use IP addresses instead of dns names. But, I don't have to deal with dynamic ip changes of any device either. ___ --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] Sipura 3000 dialplan strings.
Ken D'Ambrosio wrote: I'm trying to set up a dialplan that dials via PSTN for: All eight-digit calls that start with 9 All 911 calls All calls that start with 424 (the local exchange) I haven't tested 911 -- for obvious reasons. I may do so after I feel more confident. I've got the starts-with-9 working fine. But the local exchange stuff isn't working, and I'm confused. Here's a snippet of my dialplan: [lots deleted]|9,:xxx :@gw0|424 :@gw0) It does dial 424 numbers, but they go straight through SIP. Any suggestions? Get rid of the spaces in that dialplan. Not sure, but I don't think you want the coma in 9,: either. Try this: 9:xxx:@gw0|424:@gw0|[2-9]11:@gw0) Note the last piece supports 911, 411, 611, etc, if that's what you want. I use something like that dialplan in a spa3k, but default all outgoing calls to gw0 (fxo port) unless the user precedes the number with an 8 like this: (8:.:@gw1| If the user dials 8+something, send the call via asterisk. ___ --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: Zaptel install - Fedora Core 5
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I did yum update last week and here is my current kernel: I had no problem at all with zaptel. I am only using TDM400P though, in case that matters. Hi Anto! The thing is that I can't rely on yum update for asterisk installation. I would like something that will work like this: I install FC5 from CD/DVD, install RPM's that I need from my ftp server or from CD, install zaptel, libpri, asterisk... So, I need to download rpm's that will allow me to install zaptel/libpri/asterisk without using yum update (I need to make all installations the same). Why bother with the rpm's? After you have a working OS, why not follow the suggestions on www.asterisk.org/download and do: cd /usr/src svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2 svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2 Then, as patches and fixes are committed to the v1.2 release, you can simply do a make update, recompile, and you're always current. You're never waiting around for someone to create an rpm, etc, that way. If you want to create a simple backout plan (for asterisk, as an example), do something like this (on RH/FC systems): mv /usr/lib/asterisk/modules/* /usr/lib/asterisk/modules/backup and execute that before doing the make update. ___ --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] Linksys SPA-941 Message Waiting Indicator
voiplist wrote: Greetings.. I have a few Linksys SPA-941 IP phones running the latest firmware 4.1.12(a). I tried turning on the Message Waiting indicator but it doesn't seem to work correctly for me. This phone is connecting to Asterisk 1.24 running Realtime. Not sure if it matters but rtcachefriends=yes is set. Basically, as soon as I turn the item labeled Message Waiting to yes the red light turns on on my phone, I get stutter tone and little envelope icons show up on my phone. Doesn't matter if I have voicemail or not. I have tried filling in Mailbox ID: and VoiceMailServer: with various things but nothing seems to help. Any ideas? Would be nice to use this feature as we have with other Sipura products in the past. Sounds like a config problem with asterisk. The 941/942's here worked just fine right out of the box. ___ --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] Linksys SPA-3102
Inline... Barry D. Hassler wrote: On Sat, 2006-08-19 at 00:12 -0500, Rich Adamson wrote: Barry D. Hassler wrote: Any further experience with the 3102? I'm looking for a solution to connect 2 CO lines and a set of 2-line phones to my asterisk server (along with a bunch of SIP phones). Would 2 of these work well for that? Hopefully no echo problems! That would kill this project? I'm still searching for one peice of hardware that would have 2 FXO and 2 FXO ports on it, but haven't stumbled on it yet The only way to know for sure whether the spa will provide reasonable service is to try it on the actual pstn line to be used. There is no other way for anyone to tell you anything different. The quality of the audio echo is 100% dependent on the exact pstn line characteristics, etc. Two other reasonable alternatives are: Digium TDM card with two fxo and two fxs modules. Sangoma A200d card with one fxo module (has two lines on it) and one fxs module (has two lines on it). Yes, this would definitely work. J'm trying to avoid running phone lines the distance between the existing demarc and the asterisk server :-) But, it would probably be the best solution. Actually, as I look further, I don't think the 3102 will work in this environment at all for what I want. (incoming FXO calls go to Asterisk server, and through its dial plan, which might include ringing the analog phones on the FXS ports). That has worked just fine, even with the old spa3000. But, echo and dtmf talk-off could be an issue. I did find AudioCodes MediaPack MP-114, which has exactly the configuration I want (2 FXO, 2 FXS), but have to be registered to get any documentation on the units. Don't know anything about the MP-114 at all. If that were a solid box, I'd have to guess you'd see lots of folks recommending them. That's not the case over the last three years, so would have to guess they might have the same echo issues as the spa3k. (translated: poor to acceptable echo cancellation likely due to under-horsepowered box with software EC. 2 Grandstream HT-488's looks like a possibility too The HT-488's are total crap. The only reason that box has an fxo port is for failures, cutting the fxs directly to the fxo. Also poor echo cancellation. I think I'll just run the phone lines into the server, and add more cards there :-) Better choice. However, extending the fxs function to distant locations with an sip adapter certainly is feasible. The problem with almost all of the small/inexpensive sip adapters that have fxo ports is their software echo cancelers have rather narrow operating limits. Any calls that exceed the limits will incur echo, and that can vary from one call to another. ___ --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] Linksys SPA-3102
Barry D. Hassler wrote: Any further experience with the 3102? I'm looking for a solution to connect 2 CO lines and a set of 2-line phones to my asterisk server (along with a bunch of SIP phones). Would 2 of these work well for that? Hopefully no echo problems! That would kill this project? I'm still searching for one peice of hardware that would have 2 FXO and 2 FXO ports on it, but haven't stumbled on it yet The only way to know for sure whether the spa will provide reasonable service is to try it on the actual pstn line to be used. There is no other way for anyone to tell you anything different. The quality of the audio echo is 100% dependent on the exact pstn line characteristics, etc. Two other reasonable alternatives are: Digium TDM card with two fxo and two fxs modules. Sangoma A200d card with one fxo module (has two lines on it) and one fxs module (has two lines on it). The only way to know for sure whether the TDM card will provide reasonable service is to try it on the actual pstn lines, just exactly like the spa box. Same issues as the spa; some pstn lines work fine, others have echo that nags users. The Sangoma card with h/w echo canceler just plain works, but is probably the more expensive of the alternatives. The Mediatrix 1204 seems to have excellent echo characteristics, but it is likely the most expensive approach for small systems. ___ --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] Dial statement problem
Need a little assist by someone else's eyes; mine have gone blurry.
Running v1.2.10 checked out from svn as of today.
Problem: When dial statement is executed with a timeout value and no one
answers the call, the next priority (#4) is not being executed as expected.
When an incoming pstn call arrives, the zap/4 channel properly handles
the call and sends it to the [inbound-bus-line] context. The CLI for a
sample call appears just below the following extensions.conf paste.
When the 20 second Dial() timeout occurs, step #4 is not executed.
Rather, the next dialplan entry executed is from the next context that
immediately follows. Why?
Portion of extensions.conf:
[inbound-bus-line]
exten = s,1,NoOp,${CALLERID(all)}
exten = s,2,NoOp,bus-line-step2
exten = s,3,Dial(${PHONE1}${PHONE2}|20)
exten = s,4,NoOp,bus-line-step3
exten = s,5,Goto(bus-ivr-main|s|1)
exten = s,104,NoOp,bus-line-step103
exten = s,105,Goto(bus-ivr-main|s|1)
[inbound-bus-dialin] ; goes directly to IVR and allows most dialplan dialing
include = local-extns
include = misc-extns
include = outgoing-calls
include = parkedcalls
exten = s,1,NoOp,${CALLERID(all)}
exten = s,2,NoOp,bus-dialin-step
exten = s,3,Answer
exten = s,4,Goto(bus-ivr-main|s|1)
snip
[bus-ivr-main]
exten = s,1,Wait,1
exten = s,2,NoOp,step 2
exten = s,3,Answer
exten = s,4,Set(TIMEOUT(digit)=5)
exten = s,5,Set(TIMEOUT(response)=10)
exten = s,6,NoOp,${CALLERID(all)}
exten = s,7,Background(npi-greeting) ; Thanks for calling press 1 for
snip
phoenix*CLI
-- Starting simple switch on 'Zap/4-1'
Aug 17 11:44:52 NOTICE[15342]: chan_zap.c:6073 ss_thread: Got event 2
(Ring/Answered)...
-- Executing NoOp(Zap/4-1, Adamson Richard 402432) in
new stack
-- Executing NoOp(Zap/4-1, bus-line-step2) in new stack
-- Executing Dial(Zap/4-1, SIP/3000SIP/3001|20) in new stack
-- Called 3000
-- Called 3001
-- SIP/3000-09eed5e0 is ringing
-- SIP/3001-09ef2b20 is ringing
Note: problem starts here. The GoTo in [inbound-bus-line] step #5 is
not executed. Rather, dialplan processing continues in the next context.
-- Starting simple switch on 'Zap/2-1'
-- Executing NoOp(Zap/2-1, ) in new stack
-- Executing NoOp(Zap/2-1, bus-dialin-step) in new stack
-- Executing Answer(Zap/2-1, ) in new stack
-- Executing Goto(Zap/2-1, bus-ivr-main|s|1) in new stack
-- Goto (bus-ivr-main,s,1)
-- Executing Wait(Zap/2-1, 1) in new stack
-- Executing NoOp(Zap/2-1, step 2) in new stack
Any help would be greatly appreciated.
Rich
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Re: [asterisk-users] 1.2.10 - g726 Issues
I'm running SVN-trunk-r16869M (compiled 2006-04-01) with iax trunks to five remote systems most of which are v1.2.10. No problems with any of those trunks using g726. Teliax is the only system that I've had any issues with using iax and g726. I've not tried sip to them and don't have any intentions of doing that right now. R. Cullin J. Wible wrote: Yeah, that's exactly the problem that I am having here (also switched to g729 and gsm). However, Teliax has told me that the g726 issue is with the * 1.2.10 release and as a result not an issue with their service. I'm not entirely convinced, but since we also use g726 for some of our internal phones we must support it and if it's broken in 1.2.10 then I won't upgrade. What version of * are you runing? Thanks, Cullin -Original Message- Cullin J. Wible wrote: I have hard that 1.2.10 has issues with voice quality through g726. Can anyone provide any feedback or point me in the right direction about the current status of this problem? Been using g726 between multiple * systems for some time and the quality has been very good. Recently, however, all calls via teliax.com using g726 have had very poor quality. Switching back to gsm for them cleared up the iax audio nicely. Not sure if teliax changed something or what, but had been working fine for several months. R. ___ --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] Page()
Dennis P. Clark wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug The Page application is app_page.so (located in /usr/lib/asterisk/modules on RH systems). It is present in v1.2.10 and at least at SVN-trunk-r16869M (June 4, 2006). From the CLI, do a 'show modules like page' to see if it is loaded. ___ --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
