[asterisk-users] PAP2
I know this is slightly off topic, but I was wondering if anyone can help with a problem getting my PAP2's to connect to Asterisk. I use a provisioning file, and I recently re-wrote the files for each PAP2. I had a small typo and the PAPs logged it as a corrupt file. I corrected the file, however, Line 1 on both of the PAP2's now wont register. Line 2 works fine though. I've done the " 73738", but it wont come back. Anyone know of a way to really wipe it's memory? Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2 Dialing Delay
> Possibly OT? > I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The > only issue I can't beat with it is the dial delay when calling internal > or external numbers. > > No matter what it seems to take 10 -15 seconds to actually dial. I've > altered the device removing all *xx combos and unnecessary waffle and > cut the dialplan string to (x.S0) but the problem persists. > > Anyone else seen this issue? > Have you adjusted the Interdigit Long Timer and Interdigit Short Timer in the Regional menu? Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wierd problem
>I have asterisk setup in my house and I have a SPA-3102-NA and a PAP2T. >This > is probably just me not understanding what is going on, but I was playing > around last night and I used the sip unregister command on the > CLI. I thought the boxes would re-register when their registration > interval > was up. This is not what is happening. Now the devices are failing to > register, even my softphones (yes, I was an idiot and "unregistered" them > all). I don't know how to clear this and get my "stuff" to work again. > I've > turned Asterisk off and back on, restarted the whole machine, power cycled > the devices and pulled out some hair. Nothing seems to work. Could a kind > soul help me out? > > Tim > Finally figured it out. Somethings are too simple to notice. It was merely a DNS lookup problem. (Grumble) Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wierd problem
I have asterisk setup in my house and I have a SPA-3102-NA and a PAP2T. This is probably just me not understanding what is going on, but I was playing around last night and I used the sip unregister command on the CLI. I thought the boxes would re-register when their registration interval was up. This is not what is happening. Now the devices are failing to register, even my softphones (yes, I was an idiot and "unregistered" them all). I don't know how to clear this and get my "stuff" to work again. I've turned Asterisk off and back on, restarted the whole machine, power cycled the devices and pulled out some hair. Nothing seems to work. Could a kind soul help me out? Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
- Original Message - From: "Jeff LaCoursiere" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, January 21, 2009 9:55 AM Subject: Re: [asterisk-users] PAP2T provisioning > > > On Wed, 21 Jan 2009, Stefan Schmidt wrote: > >> Tom Moore schrieb: >>> I'm not sure if this trick will work with this device, but I was able to >>> pull down a spa8000's config by connecting to: >>> http://ipaddress/admin/spacfg.xml >>> >>> Tom >>> >>> >>> >> Hello, >> >> The spacfg.xml link doesnt work on a Pap2T but you could use this link >> to get the xml config from an Spa9xx with Firmware greater thatn 5.x. >> which is the same for pap2. >> >> I?ve attached you an complete XML file of an pap2 i?ve found. >> >> best regards >> >> Steve Smith >> > > Wow, that's perfect! Thanks! > > j > I know it's pretty much a given, but don't forget to edit/remove the provisioning info. I'd hate to see someone's open device locked. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canreinvite question
Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 & 2002 are behind one firewall, and 2003 & 2004 are behind another. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA devices and CID
Quoting Jaap Winius <[EMAIL PROTECTED]>: > Quoting Tim Johnson <[EMAIL PROTECTED]>: > >> Your caller ID is probably being over-ridden by the settings in your >> sip.conf file. Remove the caller ID from your PSTN section of the >> sip.conf, and the CID should be passed on from the POTS line. > > That sounds like a good idea regardless. On the SPA3000 I've changed > the User ID to "PSTN", while the sip.conf now has the following entry: > > [4500] > ; SPA3000, PSTN line: incoming. > type=friend > host=dynamic > port=5061 > context=home-in > username=PSTN > secret=1234 > dtmfmode=rfc2833 > disallow=all > allow=ulaw > insecure=very > qualify=yes > > While still not a solution in my case, this is an improvement. CIDs > for incoming PSTN calls are now reported as "Unavailable", instead of > always being "4500". > > Thanks! > > Jaap What do you have for your "PSTN Answer Delay" (in PSTN tab)? I had to set mine between 3 to 5 to get reliable CID from the POTS line. This was for a SPA3102, not a 3000. I've never had a 3000, but everyone says they are nearly identical. Tim Johnson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA devices and CID
Quoting Jaap Winius <[EMAIL PROTECTED]>: > Hi list, > > After successfully configuring Linksys SPA3000 and SPA3102 devices as > Asterisk PSTN gateways, the only thing I can't get working is the PSTN > Caller ID. The analog and SIP phones I've used can both display CIDs > for internal calls, while the analog model also displays CIDs > correctly when attached directly to the PSTN line. However, when PSTN > calls come in via the SPA device, all I see is the SPA device CID > associated with the PSTN line; not the CID of the incoming call. > > The only SPA settings I know of that are supposed to enable the > passing on of PSTN CIDs are the "PSTN CID For VoIP CID" option (under > PSTN Line), which AFAIK must be set to "yes," and the "Caller ID > Method" (under Regional), which I must set to "ETSI DTMI With PR", or > else my analog phone will not display any CIDs when attached to the > SPA's FXS port. Yet, these settings have never led to any positive > results, despite attempts with different firmware versions on both > devices. > > Can anyone help? > > Thanks, > > Jaap Your caller ID is probably being over-ridden by the settings in your sip.conf file. Remove the caller ID from your PSTN section of the sip.conf, and the CID should be passed on from the POTS line. Tim Johnson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 registration problem
Quoting Jaap Winius <[EMAIL PROTECTED]>: > Hi list, > > After failing to get a Sipura/Linksys SPA3000, which I've configured > as a PSTN gateway, to pass on the Caller ID, I decided to try my luck > with a Linksys SPA3102 after hearing some promising stories. > Unfortunately, I've run into a completely new problem: it seems > Asterisk won't let this device register. > > I went about configuring the SPA3102 in much the same way as I did the > SPA3000 and the Linksys PAP2T. For example, in all three cases this is > the way I configured /etc/asterisk/sip.conf for Line 1: > > [4000] > type=friend > host=dynamic > context=phones-m > secret=1234 > dtmfmode=rfc2833 > disallow=all > allow=ulaw > qualify=yes > > The device is configured to register Line 1 with the SIP proxy and as > a result the command "sip show peers" would eventually say the > following: > > Name/username Host Dyn Nat ACL PortStatus > 4000/4000 192.168.1.3 D 5060OK (13 ms) > > Not so with the SPA3102, in which case I always get: > > Name/username Host Dyn Nat ACL PortStatus > 4000 (Unspecified) D 0 UNKNOWN > > After some tests, I found out that the SPA3102 is indeed trying to > register, but that Asterisk seems to be ignoring it. Using tcpdump, I > can see that registration packets are regularly being sent to the > Asterisk server (bitis): > > 15:30:49.567288 IP spa3102.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 482 > Eh% > ...xREGISTER sip:192.168.1.10 SIP/2.0 > Via: SIP/2.0/UDP 12 > 15:30:49.568390 IP spa3102.umrk.to.sip-tls > bitis.umrk.to.sip: SIP, > length: 492 > Eh%x... > ...%REGISTER sip:192.168.1.10 SIP/2.0 > Via: SIP/2.0/UDP 12 > > This sequence keeps on repeating. Also, if I change the sip.conf > settings above to "type=peer" and "host=192.168.1.3", I'll see these > messages appear on the Asterisk console: > > [Feb 27 15:17:34] NOTICE[10893]: chan_sip.c:12414 > handle_response_peerpoke: Peer '4000' is now Reachable. (7ms / 2000ms) > [Feb 27 15:17:35] ERROR[10893]: chan_sip.c:8513 register_verify: Peer > '4000' is trying to register, but not configured as host=dynamic > [Feb 27 15:17:35] NOTICE[10893]: chan_sip.c:14943 > handle_request_register: Registration from 'Margriet > ' failed for '192.168.1.3' - Peer is not > supposed to register > > If, in this case, I configure the SPA3102 not to register any of its > extensions, Asterisk will report them to be reachable and there won't > be any more errors on the console, but in actual fact the extensions > won't be available: I won't be able to call the phone attached to it > due to congestion, and if I pick up that phone to make a call, I'll > immediately hear a busy signal. > > What could be causing this situation? I'm using Asterisk 1.4.14 and > the SPA3102 has the latest firmware version: 5.1.7(GW). I should also > mention that I'm not interested in using this device's broadband > router functionality. > > Any help would be much appreciated! > > Thanks, > > Jaap I see you put a password line in your sip.conf, but I do not see a username line. Also, you might want to check the port #'s for both the Line 1 and PSTN line. I use 5060 and 5061, respectively. Hopefully this either helps, or puts you on the right track. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best ATA. Period.
Quoting SIP <[EMAIL PROTECTED]>: > Adam Moffett wrote: >> In all seriousness, my requirements were a little silly. A Cisco router >> can fail just as a netgear router can. But I think we would find Cisco >> failures to be statistically less likely. >> >> I also think we can agree that not all devices of a certain type are >> created equal. Do you have any opinions on which VoIP products are more >> likely to be consistent and reliable? >> >> > Realistically, I've had issues with every ATA I've used to SOME degree. > The Leadtek BVA series has numerous issues. I've had bizarre things > occur in all of my Linksys/Sipura adapters(2000,3000,3201) (issues with > timeouts on a lost connection, NAT traversal, etc). My Grandstream > HT486 and 488s have intermittent dialing failures. I've had a lot of > issues with the Audiocodes MPs. > > The only ATA I've NOT actually had any issues with has been my > Grandstream HT386. Granted, I have issues with its capabilities overall, > but on the whole, it's the only one that's not simply had some weird > random failure as the others have. > > Does this mean I'd recommend an HT386 as solid testing piece? Heavens > no. I'd probably recommend the Linksys SPA3102. But be aware that there > ARE issues with just about all of them, and I doubt there have even been > enough variants sold/used by everyone to merit statistical analyses. > What you're going to get for recommendations will be, at best, anecdotal. > > N. > I have two SPA3102s, and two PAP2-NAs. All of them have worked flawlessly. I have my SPA3102s on a real IP, and have tested the PAP2's from real IPs and behind nat. They do seem to "just work" after being configured. My only complaint is with Linksys/Cisco's support, or lack of it. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-3000 caller ID and KPN
Quoting Jaap Winius <[EMAIL PROTECTED]>: > Hi list, > > Hopefully, some of our Dutch members can help with this one. I'm also > based in the Netherlands and am using a Sipura (Linksys) SPA-3000 > (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test > system. It works fine, except that the Called ID (CID) is not working. > I'm aware that KPN (our local telco) requires a separate subscription > to activate CID on POTS lines and I've confirmed that is working. Yet, > I've not been able to get the SPA-3000 to pas the CID on to Asterisk, > and I know of no relevant SPA-3000 settings for doing this other than: > > * Regional > Miscellaneous > Caller ID Method: ETSI DTMF with PR > > * PSTN Line > PSTN-To-VoIP Gateway Setup > PSTN CID For VoIP CID: yes > > As some have suggested, I've also set the Regional / Miscellaneous / > Caller ID FSK Standard: to 'bell 202', but this seems like nonsense to > me as it should not make a difference once you've selected a DTMF CID > method. > > I've also experimented with increasing the answer and ring-through > delays, but this makes no difference. I've been told that this is > because KPN always sends the CID on ahead of the rest of the call to > begin with. > > Could it be that I'm missing something? > > Thanks, > > Jaap Hi Jaap, I have a SPA3102 which is supposed to be similar. Make sure you leave the PSTN --> Subscriber Information --> Display Name blank. Also, in your sip.conf file, do not specify any "callerid=" value. This probably isn't your issue, but I thought just in case... Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 without licence
Hello all, I am running Asterisk 1.4.17. I have 2 Linksys SPA3102's and one PAP2-NA (I have a second on order). They have G729a built into them. This is supposed to be compatable with G729. I was trying to have them use that codec when they talk to each other, but it seems they always switch to alaw or ulaw (depending on my sip.conf file). Shouldn't they be able to use G729a in passthru when connecting to each other? My PAP2 and one SPA3102 are on the LAN, but I have one remote 3102 that I'd like to use lower bandwidth codec with. disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm Tim Johnson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need sample configuration files for sipura/linksys ata
I have emailed Linksys about this, and they have not answered. I have figured out much of how to do this, despite Linksys not being any help. My only remaining issue is how to configure the PSTN line on a SPA3102. Try a file like this (example info included); Yes yes 30 1200 300 3600 yes yes no yes Notice the . I don't have my PAP2-NA yet, but according to the information I found, line two would be . The lines have a _1_ or _2_ etc etc. Remember to end each line as above. I have mine setup so it downloads the first profile () from tftp. That then loads the 3102 with the http site for the rest of the configuration. You can use variables in your URL/TFTP line as well. A $MA will send the MAC address of the adapter, and $PSN is the model number (such as "3102") and $SN is the devices serial number. If you figure out how to specify settings for the PSTN line, please share it with the list. Tim Johnson Quoting Gopal krishnan <[EMAIL PROTECTED]>: > Hi, > > Try this > > http://www.kcip.com/support/pap2uk.html > > On Jan 25, 2008 4:18 PM, Rizwan Hisham <[EMAIL PROTECTED]> wrote: > >> Hi all, >> i need sample xml configuration files for linksys pap2, linksys pap-2t, >> sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are >> linksys/sipura products. So if anyone has these sample files then plz share. >> >> >> -- >> Best Regards >> Rizwan Hisham >> Software Engineer >> Axvoice Inc. >> www.axvoice.com >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound dialing
Hi Drew. Thanks for the tips. My Line 1 works as I'd like it to, and I could be
wrong, but I don't think changing the dialplan there will help. I really just
want to be able to dial local phone calls (7 digits) and have it go out the
SPA3102, without having to dial twice. This is a snip what I have so far.
extentions.conf
exten => _NXX,1,Dial(SIP/201/${EXTEN},20)
exten => _NXX,2,Hangup
sip.conf
[201]
type=friend
username=x
secret=x
host=dynamic
context=sip
nat=yes
canreinvite=yes
qualify=yes
subscribecontext=localextensions
dtmfmode=rfc2833
vmexten=voicemail
disallow=all
allow=ulaw
allow=gsm
On the SPA (in the "PSTN Line" tab)
Dial Plan 1: ()
Dial Plan 2: S0<:255>
DialPlan 1 is just what I have for now
DialPlan 2 is my extention 255, with a PSTN call comes in it rings my SIP phone.
I set my VoIP Answer Delay to 0 (from 1, I also tried 5 some time ago) and I
set the SPA To PSTN Gain to 5 and now 15.
With things the way I have them now, when I dial a local number, I get a single
DTMF tone on the phoneline, not sure what digit it is.
Tim
- Original Message -
From: Drew Gibson
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, August 07, 2007 5:55 PM
Subject: Re: [asterisk-users] Outbound dialing
Tim,
If the Asterisk stuff below doesn't fix it, try the docs at
http://www.jmgtechnology.com.au/spa_3000_guide.pdf
Ensure you enable VoIP to PSTN gateway mode and that "PSTN Line" is
registered with Asterisk. This is probably OK as you appear to get dialtone
back from the SPA. If you are calling from the phone on "Line 1", make all
calls go through Asterisk. See above docs for details.
In case you are dialing from a phone on "Line 1", here is the "Line 1"
dialplan from my home SPA3102...
(*xx|[3469]11|0|00|[29]x|1xxx[2-9]xx|2[01]x|50[01]|.)
I can't remember if that is default or if I tweaked it. Works in Ontario.
If that is OK, try increasing the gain "SPA to PSTN". If the gain is too low,
the DTMF may not be recognised by the CO. I found this out whilst
troubleshooting echo problems.
regards,
Drew
Nicholas Blasgen wrote:
Not specific to the SPA3102, but just normal outbound dialing is as follows:
exten => _1NXXNXX,1,Dial(//${EXTEN})
or if you want to require people to dial 9, then:
exten => _91NXXNXX,1,Dial(//${EXTEN})
or if you're like me and you're used to a cell phone and don't like dialing
the 1:
exten => _NXXNXX,1,Dial(//1${EXTEN})
On 8/7/07, Tim Johnson <[EMAIL PROTECTED]> wrote:
Hello all. I am just getting back into Asterisk and I am setting up my
Linksys SPA3102. I have incoming calls working fine, as is the phone
plugged into the unit. My problem is I cannot get the SPA3102 to dial
a phone number automatically. I can call the extention of the PSTN and
I get a second dialtone, and I can then manually dial. I'd like to be
able to have Asterisk pass the number I dialed to the SPA and have it
dialout. I've played with dialplans on the SPA I've found during my
googling, but I think it might be something I am missing in my
extentions.conf file. Any ideas?
Tim
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/Nick
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--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
--
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[asterisk-users] Outbound dialing
Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the number I dialed to the SPA and have it dialout. I've played with dialplans on the SPA I've found during my googling, but I think it might be something I am missing in my extentions.conf file. Any ideas? Tim ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gnet VP168S
Has anyone used a Gnet VP168S with Asterisk? I've been testing with softphones, and this will be my first attempt at using a hardware product to connect a standard POTS telephone. The limited specs I found online suggest it should work (SIP one FXS port and one PSTN "Fall back" port, but like I said, this is my first attempt at using hardward (and I've only been playing with Asterisk/softphones for 5-6 days). Any input would be appreciated. Tim Johnson - This message was sent using the fabrysociety.org Webmail. For more Information please visit ; http://www.fabrysociety.org ___ --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
