[Asterisk-Users] SIP quality monitoring
Is there a way to get a report from Asterisk on the quality metrics (packet loss, delay, jitter) of at least the inbound component of a SIP call? Thanks James ___ --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] Nokai E60 and E61 , working fine with Asterisk , with new access points
Hi Was able to communicate clearly with e60 and E61 with asterisk with new access point , even though the access point security setting was of opennetworks , the previous one was of WEP , I feel this was a major hurdle in communication , now I can clearly accept and make calls using Nokia E60 and E61 devices Next I will be trying to find out how to make this device work , with WEP enabled security will be posting about the findings Thanks Joseph John Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --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] ADSL modem, TDM400P, zaptel and not hanging up
Nick Chalk wrote: [EMAIL PROTECTED] wrote: I've got speedtouch ones at home, here I've got a Zoom one and a Dlink one I can try, It will be a bit of a botch-job, atm. I'm using one of those nice ones that plug into the front of an NTE-5 (so I can punch the cables straight in). An NTE-2000? Those are reckoned to have pretty good filters. It's probably worth trying another filter, in case there's a fault with your current one. Nick. The ones I had to hand didn't work. Could try swapping which FXO port is in use, replacing the card with an ATA is out of the question though, since modem calls will need to be made down the same line. (albeit low-speed ones). ___ --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] Question setting up a bat phone extension.
James Harper wrote: Easy to do on the Linksys PAP2, if that helps. The functionality probably depends on the make and model of the phone... maybe if you gave those details as well? James Fantastic, this may solve the problem In the mail I've just posted (which hasnt' appeared yet). I don't remember seeing this function in a PAP2, is it fairly trivial to set up? Will I need a particular firmware version? ___ --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] OLD PA system.
I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is no on-hook state in the whole setup. Obviously If I just connect the input to a port on an ATA, I'll just get a dialtone played through the speakers. Can anyone think of a way I can attach it, so that people can call an extension that will play through the speakers? The nearest thing I can think of, is getting the * box to call it as an extension plugged into an ATA, then while it's calling plug the line in and transfer it to a non-announcing conference (or similar). This is an undesirable approach for many reasons. (mostly the requirement to set it all up again in the event of a server reset). I have a small pile of ATAs I can play with (well, there's already an atcom 468 in there with a spare port, there's a spare PAP2, a spare Cisco 186 or I could borrow an SPA3k from my home setup) If they'd be of any help. I originally thought about using the sound card, till I saw what was already in place. (The client doesn't like spending money). ___ --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: VGSM Trouble: Kind people, help me please...
Thanks a lot for responding. I did what you recomended, and it works now. At least I can make simple calls out. Did not try the incoming part though. Now it is still unclear : - how to make the Dial application choose the first available channel? the easiest (for you) is installing freepbx (or amportal) set up the trunks (i mean all the four channels) and add all the trunks to outbound routing - or how to get CID out of the interface? Does it set the Global variables as it is in Zaptel? it works right. - ...and a lot of stuff alike, as it usually happens with newly developing project... you R welcome -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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] Callback Application: Suggestions Please.
Keyboardot ragadtam, hogy va'laszoljak Tigran Kocharyan osszedobalt bytejaira: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the Outgoing number through another channel (SIP/IAX/ZAP) and bridges the call. 5. After the completion, I should see the Disconnect Reason and the Duration for each leg of the call. The first two steps are quite evident. Now the trick comes on step 3. How to Dial out a number and listen for DTMF tones? After this, maybe park the call, or send it to conference playback(hellomate) DISA(1234|context) -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.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] Question setting up a bat phone extension.
James Harper wrote: Easy to do on the Linksys PAP2, if that helps. The functionality probably depends on the make and model of the phone... maybe if you gave those details as well? James Fantastic, this may solve the problem In the mail I've just posted (which hasnt' appeared yet). I don't remember seeing this function in a PAP2, is it fairly trivial to set up? My dialplan in the pap2 is: (:0S0) Which causes it to dial a '0' to asterisk as soon as I gets picked up. In my asterisk dialplan it then does a DISA to another context, which means Asterisk is doing all the dialplan stuff. For what I want in a dialplan, I could have configured it in the pap2 but I didn't want to learn it. I think I'm at that age where everything new I learn means something else gets overwritten :) My extensions.conf looks like: [pap2_in] exten = 0,1,Answer exten = 0,n,DISA(no-password|internal) exten = t,1,Congestion() Ideally I would have liked the pap2 to have done the same as 'immediate' when talking about fxo, capi, misdn, etc, but I couldn't get it to automatically dial nothing. A '0' was the best I could do. If anyone knows how to put it into immediate mode to come into asterisk as an 's' extension, I'd love to hear about it! Will I need a particular firmware version? Not sure. I'm using 3.1.3(LS), but if I remember correctly I had this working on a prior firmware too... Good luck. James ___ --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] Question setting up a bat phone extension.
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote: Ideally I would have liked the pap2 to have done the same as 'immediate' when talking about fxo, capi, misdn, etc, but I couldn't get it to automatically dial nothing. A '0' was the best I could do. If anyone knows how to put it into immediate mode to come into asterisk as an 's' extension, I'd love to hear about it! sip targets arent limited to numerics, have you tried to dial an 's' instead of '0'? That is valid in sip, I just dont know if the pap2 supports it. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com we pay you to terminate calls with us! signature.asc Description: This is a digitally signed message part ___ --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] to china: good voip service providers?
Dear list, I've been looking for a voip service provider with inexpensive and high- quality call service to China. However, the providers I've tried (voipjet, exgn, voxee) all have long to super-long latencies on calls to China. Has anyone found a service with good connections to China? Please share! John PS: I'm trying out this 'didww.com' service for the first time for Chinese DIDs. The website is terrible and makes setup confusing, and on first try, it sounds like I'm getting some dropped packets. However, the service was very fast to set up (once I figured out how to add the Asterisk username/passwd on the website), and I do now have a semi- working DID in Shenzhen! (Am having trouble passing DTMF, though) This is the second service I've found with inexpensive, asterisk- compatible Chinese DIDs after ctcvoip, which is extremely unreliable (as in: simply stops working for days on end). ___ --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] asterisk-1.2.9.1
hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error [pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! can anyone help me i have redhat linux enterprise zaptel version 1.2.6 libpri version 1.2.3 what am i missing here? ___ --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] Question setting up a bat phone extension.
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote: Ideally I would have liked the pap2 to have done the same as 'immediate' when talking about fxo, capi, misdn, etc, but I couldn't get it to automatically dial nothing. A '0' was the best I could do. If anyone knows how to put it into immediate mode to come into asterisk as an 's' extension, I'd love to hear about it! sip targets arent limited to numerics, have you tried to dial an 's' instead of '0'? That is valid in sip, I just dont know if the pap2 supports it. Hey... look at that ... it works! Cool :) So... asterisk can't tell the difference between 's' for 'no extension dialled', and when 's' was actually the name of the extension dialled... is this the expected behaviour? More importantly, is this a security risk in any way? I can't think of a situation where it would be... Even more importantly, could this change in a future version? Thanks James ___ --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] Question setting up a bat phone extension.
James Harper wrote: So... asterisk can't tell the difference between 's' for 'no extension dialled', and when 's' was actually the name of the extension dialled... is this the expected behaviour? I surely hope so, you can refer to it as such in the extensions.conf as well (with goto etc.) also works with t, h, i and presumably T and o. ___ --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-1.2.9.1
amna saleem wrote: hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error [pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! can anyone help me i have redhat linux enterprise zaptel version 1.2.6 libpri version 1.2.3 what am i missing here? I know it sounds daft, but could it be a module that was compiled as part of a previous install of asterisk and wasn't overwritten/deleted and is being autoloaded (or directly loaded)? It's not a standard module (afaik) anymore, and certainly hasn't been compiled with the versions I'm running. If it's for a particular card, you may need to recompile the module yourself from their driver source. If you don't know what it's used for, I'd move it out of /usr/lib/asterisk/modules, and see what happens/doesn't happen. Usually when you update asterisk, at the end of the make-install it will give you a list of modules that are in the modules directory that it hasn't placed there, if it does that they are usually worth looking at. ___ --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] OLD PA system.
Thomas Kenyon wrote: I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is Does the connection use 2 screws for analog inputs? If this is the case, you could get a cheap Grand Stream BT102 and pull the speaker leads off and connect it to that box. The GS can be setup to auto answer. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OLD PA system.
Doug Lytle wrote: Thomas Kenyon wrote: I need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is Does the connection use 2 screws for analog inputs? Yup. If this is the case, you could get a cheap Grand Stream BT102 and pull the speaker leads off and connect it to that box. The GS can be setup to auto answer. Doug I'm going to try the suggestion in the Bat Phone thread above, bringing one of the PAP2s out of retirement. Wish me luck. ___ --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] Cisco router and 488 Not acceptable here messages
Are there any known problems with Cisco routers (Cisco 837) and SIP sessions? I have been trying to track down a problem for about 3 hours now and I think the Cisco router is the culprit!!! I keep getting 488 Not acceptable here messages, which are apparently normally the message you get when a common codec can't be found. I'm also getting chan_sip.c:3434 process_sdp: Insufficient information for SDP (m = '', c = '') messages, which is strange because the m and c attributes are definitely there. When I looked closer, they are received on the first INVITE, then asterisk says 'proxy auth required', but the INVITE packet with the proxy auth doesn't have the attributes. A tcpdump on the asterisk server confirms this. But, when I do a ethereal dump on my PC where I'm running SJphone, the attributes are there in the packet. So something is futzing with my packets, and screwing them up. There is a Cisco router on my end of the link, so I'm suspecting that! Any suggestions? Thanks James ___ --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] hook flash call transfer
I am trying to use hook flash to transfer a call but I want the recording on the line I transfer to to start after I hang up. In other words if I receive a call and want to transfer it to VM or to a recording, I want to be able to flash the hook, dial the extension, and hang up. But I do not want the recording/vm message to satrt until the call is actually transfered. Is this possible? My work around it to insert a wait in the beginning of the contect I am transferring to. Is there a cleaner way? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --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 and 488 Not acceptable here messages
Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Monday, 12 June 2006 00:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco router and 488 Not acceptable here messages Are there any known problems with Cisco routers (Cisco 837) and SIP sessions? I have been trying to track down a problem for about 3 hours now and I think the Cisco router is the culprit!!! I keep getting 488 Not acceptable here messages, which are apparently normally the message you get when a common codec can't be found. I'm also getting chan_sip.c:3434 process_sdp: Insufficient information for SDP (m = '', c = '') messages, which is strange because the m and c attributes are definitely there. When I looked closer, they are received on the first INVITE, then asterisk says 'proxy auth required', but the INVITE packet with the proxy auth doesn't have the attributes. A tcpdump on the asterisk server confirms this. But, when I do a ethereal dump on my PC where I'm running SJphone, the attributes are there in the packet. So something is futzing with my packets, and screwing them up. There is a Cisco router on my end of the link, so I'm suspecting that! Any suggestions? Thanks James ___ --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] Virtual PBX Billing and Management Software
Destar[1] has recentely included Virtual PBX features inside it's main funcionality (right now you have to download the trunk developement branch to get it), but it would be availabe on version 0.2 coming soon in a few weeks. [1] http://destar.berlios.de/jmaczOn 6/9/06, William Piper [EMAIL PROTECTED] wrote:checkout http://www.asterisk2billing.org bp On 6/9/06, Daniel Salama [EMAIL PROTECTED] wrote: Is there any open source software capable of managing Asterisk tooffer Virtual PBX services to multiple clients, including billing? Or is there a combination of open source initiatives that offer this?Thanks,Daniel___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- I know what hell is. It´s not lakes of burning oil, or brimstone and devils poking you in the ass with pitchforks. Hell is not knowing. Ted McKeever. ___ --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 PAP2T-NA - call goes through but phone doesn't ring
I had a bunch of PSP2-NA devices with firmware 3.x that did that. Downgrading to 2.0.13 solved the problem. Others said that the last 3.x would do also, but after putting out hundreds of PAP2 with 2.x and they all working rock solid, I'm not willing to switch to 3.x until I have tested it enough (which I havent). The problematic 3.x firmwares would also make some of the PAP2 reboot when receiving a call. As soon as you call the extension, it flashes red the power led and reboot (you can see it grabbing a new IP on the network). This is also solved when downgrading. Interestingly enough, some of them work well with that same firmware. I have not yet made a relation (maybe hardware version, I don't know) andre On 6/8/06, James Moore [EMAIL PROTECTED] wrote: I'm trying out a Linksys PAP2T-NA. Calling out works great, no problems there. Calling in, though, the phone doesn't ring. Caller ID shows up, I can pick up the phone, and the call is connected, but no ring. I've tried it on two analog phones, same behavior. Suggestions? Asterisk SVN-branch-1.2-r31555. - James Moore ___ --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 -- Andre Ruiz [EMAIL PROTECTED] Curitiba, PR, Brasil ___ --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] Callback Application: Suggestions Please.
I guess I've found some good references on how to accomplish this: http://voxilla.com/PNphpBB2-viewtopic-t-6320-sid-11997b0cebea526d7a7562f38c0fd595.html http://nerdvittles.com/index.php?p=73 Thanks for the hint though. Woodoo People .pGa! wrote: Keyboardot ragadtam, hogy va'laszoljak Tigran Kocharyan osszedobalt bytejaira:" 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the Outgoing number through another channel (SIP/IAX/ZAP) and bridges the call. 5. After the completion, I should see the Disconnect Reason and the Duration for each leg of the call. The first two steps are quite evident. Now the trick comes on step 3. How to Dial out a number and listen for DTMF tones? After this, maybe park the call, or send it to conference playback(hellomate) DISA(1234|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] Xorcom Rapid
Tzafrir Cohen a écrit :
I'm still not hapy with that as a default. It should provide you a basis
for manual editing at this stage. But I wonder what else could the
script configured there differently. Are those sane defaults for BRI on
France?
I've modified zaptel-channels.conf file , because, nothing happen when i
call from an external phone inside the company.
It's my problem, i don't know how name the QuadBRI interface, and how to
use it in extensions files
Do you hace some samples to give me, or explain me how i can detect the
name to use?
Best regards,
Olivier Saulnier
# Global data
loadzone= fr
defaultzone= fr
zaptel-channels.conf:
; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
; Span 1: ztqoz/2/1 quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
; Span 2: ztqoz/2/2 quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
; Span 3: ztqoz/2/3 quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 7-8
; Span 4: ztqoz/2/4 quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 10-11
extensions.conf:
[general]
static=yes
; we don't want asterisk to write the configuration, as it will write
; everything to a single file
writeprotect=yes
[globals]
#include extensions-defs.conf
; another #include. This one includes complete contetexts.
; What happens if a section that has existed is re-added?
;
; Currently Asterisk ignores the new section. And thus is is very simple
; to override existing extensions. However nobody guarantees that the
; configurations will be paserd the same way in the future. This is
intended
; for immediate hacks and for long-run system breakage.
#include extensions.d/*.conf
; Basically you should not edit this file to add new stuff: add/edit
; files in extensions.d/ instead. Fr instance: to add an IVR: look at
; extensions.d/ivr.conf and later on 'include = ivr' instead of
; 'include =phone'
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Device(s) to ring
; ${ARG2} - flags for Dial: if empty: tr. pass '-' for no flags.
; ${ARG3} - voicemail box. If empty: use the extension number.
exten = s,1,SetVar(VMBOX=${MACRO_EXTEN}); default for VMBOX, if no ARG3
exten = s,2,GotoIf($[${LEN(${ARG3})} = 0]?4)
exten = s,3,SetVar(VMBOX=${ARG3})
; Ring the interface, 20 seconds maximum
exten = s,4,SetVar(FLAGS=r)
; why 'x'? see bourne shell 101
exten = s,5,GotoIf($[ x${ARG2} = x- ]?7); '-' as the 'flags' argument
exten = s,6,SetVar(FLAGS=${ARG2})
exten = s,7,Dial(${ARG1},20,${ARG2})
; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s,8,Goto(s-${DIALSTATUS},1)
; If unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,1,Voicemail(u${VMBOX})
; If they press #, return to start
exten = s-NOANSWER,2,Goto(${MACRO_CONTEXT},s,1)
; If busy, send to voicemail w/ busy announce
exten = s-BUSY,1,Voicemail(b${VMBOX})
; If they press #, return to start
exten = s-BUSY,2,Goto(${MACRO_CONTEXT},s,1)
; Treat anything
exten = _s-.,1,Goto(s-NOANSWER,1)
;
; You may want to improve this one
;
[macro-stdmeetme]
exten = s,1,MeetMe(${MACRO_EXTEN},M)
[macro-dialout]
;
; a macro for setting up a trunk
; usage:
;
; Arguments:
;
; ARG1: trunk channels: a ''-separated list of channels
; ARG2: number: the number to dial.
;
; Example:
;
; exten = _9.,Macro(dialout,Zap/1Zap2,${EXTEN:1})
;
exten = s,1,ChanIsAvail(${ARG1}); use
exten = s,102,Goto(s-CHANUNAVAIL,1) ; this indicates that all lines
exten = s,2,SetVar(DIALLINE=${AVAILORIGCHAN})
exten = s,3,Goto(start,1) ;
include = trunk-macros-common
[macro-trunksip]
;
; a macro for setting up a trunk
; usage:
;
; Arguments:
;
; ARG1: trunk channel: a *single* channel name: SIP/peer, IAX2/peer
; Does this work for OH323?
; ARG2: number: the number to dial.
; ARG3 (optional): maximal number of calls allowed in this trunk.
; If not given: unlimited.
;
; Example:
;
; exten = _9.,Macro(Zap/1Zap2,${EXTEN:1})
;
exten = s,1,GotoIf($[${ARG3} = ]?6)
; The group name is the sip/iax peer
exten = s,2,Cut(GROUPNAME,ARG1,,1); leave only the first target
exten = s,3,Cut(GROUPNAME,GROUPNAME,/,2); extract peer name
exten = s,4,SetGroup(${GROUPNAME})
exten = s,5,CheckGroup(${ARG3})
exten = s,106,Goto(s-CHANUNAVAIL,1)
exten = s,6,SetVar(DIALLINE=${ARG1})
exten = s,7,Goto(start,1)
include = trunk-macros-common
[trunk-macros-common]
;
; a macro for setting up a trunk
; usage:
;
; Arguments:
;
; DIALLINE: trunk channels: The channel
Re: [Asterisk-Users] Xorcom Rapid
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN] ; for in coming calls - defin in zapata.conf
exten = s,1,Dial(IAX2/300,20)
exten = s,2,Voicemail, u300)
[INTERNAL] ; for internal AND outgoing call - actually just outgoing calls
exten = _0.,1,Dial(ZAP/g1/${EXTEN:1})
For hardware, how can i know on which interface is connected my ISDN line??
For outgoing call, i name the channel ZAP/1 in extensions.conf file, but
i dont know if it's correct.
And i always have the message timeout, but no rule 't' in context
What's mean??
There is no extension named t in that context to handle timeouts.
Your dialplan reads:
[PSTN]
exten = 1,1,Dial (IAX2/300,20)
exten = s,2,Voicemail, u300)
So no timeout action is specified. Ignore it if you don't just want to
have the call disconnected on timeout without taking any other action.
I'm not sure if the space after Dial is legal. I figure it may be the
source to your problem. Do you get an error in the CLI when reloading?
Before reloading:
set verbose 1
to see only the relevant warnings.
I have the same message!
Do you know how i can stop messages from qozap (they fill the screen
either asterisk is down!!!)
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage Diamecans
Bel Air
03410 St Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
___
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Re: [Asterisk-Users] Nokai E60 and E61 , working fine with Asterisk , with new access points
On Jun 11, 2006, at 2:32 AM, John Joseph wrote: Hi Was able to communicate clearly with e60 and E61 with asterisk with new access point , even though the access point security setting was of “opennetworks” , the previous one was of “WEP” , I feel this was a major hurdle in communication , now I can clearly accept and make calls using Nokia E60 and E61 devices Next I will be trying to find out how to make this device work , with “WEP” enabled security will be posting about the findings How does the speakerphone function sound? Does the phone support WPA security? Is your wireless access point on the same LAN as the asterisk box? Thanks for the info. Marty ___ --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 and 488 Not acceptable here messages
On Jun 11, 2006, at 8:15 AM, James Harper wrote: Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. If you are behind a NAT perhaps two SIP devices are both trying to use 5060? Just a thought. Marty ___ --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-1.2.9.1
Hi Amna, Make a test In the archive modules.conf places the following line: noload = pbx_wilcalu.soStopasterisk and initiates asterisk again. It mustresolv its problem. I wait to have helped. Greetings Josué 2006/6/11, Thomas Kenyon [EMAIL PROTECTED]: amna saleem wrote: hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error [pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! can anyone help me i have redhat linux enterprise zaptel version 1.2.6 libpri version 1.2.3 what am i missing here?I know it sounds daft, but could it be a module that was compiled aspart of a previous install of asterisk and wasn't overwritten/deletedand is being autoloaded (or directly loaded)? It's not a standard module (afaik) anymore, and certainly hasn't beencompiled with the versions I'm running.If it's for a particular card, you may need to recompile the moduleyourself from their driver source. If you don't know what it's used for, I'd move it out of/usr/lib/asterisk/modules, and see what happens/doesn't happen.Usually when you update asterisk, at the end of the make-install it willgive you a list of modules that are in the modules directory that it hasn't placed there, if it does that they are usually worth looking at.___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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 router and 488 Not acceptable here messages
James Harper wrote: Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. James You probably have are behind NAT and your NAT device has a SIP ALG. Changing the port disables the ALG. The ALG is broken. -- Andres Technical Support http://www.telesip.net ___ --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] Re: Nokai E60 and E61 , working fine with Asterisk , with new access points
John Joseph wrote: Was able to communicate clearly with e60 and E61 with asterisk with new access point [..] Could you please post some details (or even better: write them in some sort of Wiki) on the configuration you did on the Nokia? I'm thinking about buying a Nokia E60 but after a short web search there seem to be some problems about the correct configuration of the phone. Greetings, Markus ___ --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] quad t1 / 1U rack server combos
Colin Anderson wrote: C'mon guys! Certify a few current model servers and be done with it. Problem is, certification is a moving target and can become invalid with something as simple as a BIOS change by the manufacturer. Now that the barrier to entry to changing a design is almost nil, manufacturers love to screw around with designs to save a few bucks. I have seen two identical boxes, labelled as such by the manufacturer, bought in the same time period, but with different guts. Digium would wind up with egg on their faces by certifying a system, then 90 days later after everyone buys it, finds out that some subtle change by the manufacturer has destabilized the config. I agree it is frustrating as hell, but this is the price we pay. Would you rather buy a Mitel for 10X the $$$? Maybe in some circumstances, it is worth it. lman/listinfo/asterisk-users I bought two HP DL380s at the exact same time from CDW. I used the first one to build an image to transfer to the other system. When I booted the second system, kudzu reported different NICs. So, yes, next to impossible to certify any hardware without the hardware manufacturer's help. ___ --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 and 488 Not acceptable heremessages
On Jun 11, 2006, at 8:15 AM, James Harper wrote: Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. If you are behind a NAT perhaps two SIP devices are both trying to use 5060? Packets are getting out, but one critical packet has had the body of the INVITE removed, so Asterisk at the other end thinks that no audio codecs have been proposed. James ___ --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] Changing RO vars like SRC
Guys, is there a way to set CDR vards like SRC, I tried using set but asterisk complains they are RO vars. What Im trying to do is a small way to let users make calls from someone elses extension but auth using a password and seitch credential to their own so the call appears on CDR as made from their extension and not the one they are actually using. Is there a way to do this and somebody has done this before? ___ --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] ISDN and DVO
I'm looking at setting up an ISDN internet service for someone, and she'd like to be able to do VoIP. The modem (230kbps serial and 2 POTS ports) you get from the ISP can do DVO (Dynamic Voice Override) where you can be online at 128kbits/sec (2 channels), but if a voice call is detected (call waiting signalled via the D channel I guess) or if you want to make a call out, one of the channels drops to be used for voice, leaving the data at 64kbps. When the voice call finishes, the modem goes back up to 128k. Has anyone configured a system with Asterisk and a BRI adapter to do the same thing for when you aren't making a VoIP call? (calls to mobiles are normally cheaper via PSTN than VoIP on the cheaper plans from what I've seen in Australia). I could also use a SPA3000 or something in line with the voice port on the modem, but having asterisk do it all would be quite nice. As usual, any and all comments appreciated! Thanks James ___ --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] SOLVED - Cisco router and 488 Not acceptable here messages
James Harper wrote: Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. James You probably have are behind NAT and your NAT device has a SIP ALG. Changing the port disables the ALG. The ALG is broken. I was going to try upgrading the IOS on the router sooner or later, but did it sooner on the basis of your comments, and it's now working! Thanks! If anyone is interested, the Cisco device in question is a Cisco 837-K9-64 ADSL modem/router. I was using IOS 12.4.7 and it was botching up the outgoing INVITE packets. Upgrading to IOS 12.4.8 solved the problem. The INVITE packet was being sent by SJphone (a soft SIP phone), so maybe something funny in there was triggering the bug in the Cisco IOS, and it would otherwise have passed SIP packets from other clients just fine. James ___ --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] JIAX status
HI, Anyone knows the current status of JIAXclient? I tried to recompile the sources available in sourceforge but they reference a old java package that I was not able to find. I tried to e-mail the author but seems that his account is no longer valid. I in need of a java IAX client that could be loaded as an applet. I know thatis a lot of viable SIP alternatives, but due to NAT/Firewall restrictions use ofIAX would be easier.Thanks in advance, -- Rubens Zupelli Filho[EMAIL PROTECTED] ___ --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] JIAX status
Rubens Zupelli Filho [EMAIL PROTECTED] writes: Anyone knows the current status of JIAXclient? I have been playing with jiaxclient 0.0.6, and it seems to mostly work if you have a working copy of the C iaxclient library. I would test iaxclient with the command-line tools that come with it and make sure that all works satisfactorily before moving on to JIAXClient, since the Java code is built on top of the C library. I tried to recompile the sources available in sourceforge but they reference a old java package that I was not able to find. What package? Scott. ___ --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] JIAX status
Scott, You are compiling in Linux or Windows? The package the java compiler is not founding is: net.sourceforge.iaxclient.jni many thanks. On 6/11/06, Scott Gifford [EMAIL PROTECTED] wrote: Rubens Zupelli Filho [EMAIL PROTECTED] writes: Anyone knows the current status of JIAXclient? I have been playing with jiaxclient 0.0.6, and it seems to mostly work if you have a working copy of the C iaxclient library. I would test iaxclient with the command-line tools that come with it and make sure that all works satisfactorily before moving on to JIAXClient, since the Java code is built on top of the C library. I tried to recompile the sources available in sourceforge but they reference a old java package that I was not able to find. What package? Scott. ___ --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 -- Rubens Zupelli Filho [EMAIL PROTECTED] ___ --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] David Choo/eServices/eSpore is overseas
I will be out of the office starting 12/06/2006 and will not return until 17/06/2006. Dear Sir / Mdm, I'm currently travelling. During this period of time, I have minimal access to internet and email. As such, please be aware that I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. For General Technical Queries, please contact Mr Tony Chew @ (65) 6842 2725, Option 2 For VoIP Technical Queries, please contact Mr Randy Khor @ (65) 9800 8468 For Sales Related Queries, please contact our Sales Hotline @ (65) 6842 2725, Option 1 Should you wish to reach me urgently, please contact me @ (65) 6842 2725, Ext - 404 instead. Alternatively, you might wish to drop me a SMS at (65) 90062645 and I will get back to you once I get 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
[Asterisk-Users] TTS engine query
Not being very happy with festival I would like ro get a better TTS engine. I looked at the listings at: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international but I would like to get user input on suggested packages for Linux. Best performance vs. cost Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --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] hangup lag causing the answering of already answered calls
I'm having the exact same problem. Please any ideas? My IP phones keep ringing after PSTN hangup or PSTN answer... for about 6 or 7 seconds.On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stopringing?It's quite a show stopper... imagine ringing a business and beinganswered by 3 different people, one after the other, all talking overthe top of each other.On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything.
Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line?exten = s,1,Dial(IAX2/carey)Nothing else? Try adding a Hangup command on the next priority and see if that helps any.exten = s,2,HangupIf you already have a Hangup command in there,
then I apologize for wasting your time. :)Undrhil--- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the
call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the Asterisk console that my phone is called twice every time there is an incoming call. Is this normal, and could it be causing this behaviour? If not, any ideas
as to what could be causing this? I can provide full debug logs and my relevant configuration if needed. Console log: -- Starting simple switch on 'Zap/4-1' -- Executing Dial("Zap/4-1", "IAX2/carey") in new stack -- Called carey -- Starting simple switch on 'Zap/4-1' -- Executing Dial("Zap/4-1", "IAX2/carey") in new stack
-- Called carey -- Call accepted by 10.0.12.102 (format ulaw) -- Format for call is ulaw -- Call accepted by 10.0.12.102 (format ulaw)-- Format for call is ulaw -- IAX2/carey-1 is ringing -- IAX2/carey-1 is ringing -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup
'Zap/4-1' -- Hungup 'IAX2/carey-1' == Spawn extension (incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stopringing?It's quite a show stopper... imagine ringing a business and beinganswered by 3 different people, one after the other, all talking overthe top of each other.On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea
wrote: Hi Undrhil, A logical idea, but unfortunately adding it didn't change anything. Two important points: (1) When I test this with just IAX endpoints, no Zap, the call is hungup immediately, (2) but the console still shows the user being called twice. So as a wild guess, maybe the console logging twice is OK, and it's my Zap configuration? * extensions.conf: [incoming] exten = s,1,Dial(IAX2/carey) exten = s,2,Hangup(IAX2/carey) * zapata.conf: [channels] usecallerid=no signalling=fxs_ks context=incoming channel = 4 * zaptel.conf loadzone=au defaultzone=au fxsks=4 * ztcfg -vv Channel 04: FXS Kewlstart (Default) (Slaves: 04) 1 channels configured. I'm from Australia so I assume the
loadzone and defaultzone is OK as per zaptel.c. Did not post iax.conf due to my SIP phones having the same behaviour, and IAX-to-IAX not exhibiting the problem. On Fri, 2006-06-09 at 04:54 +, [EMAIL PROTECTED] wrote: So, your dialplan for that incoming call is just the one line?exten = s,1,Dial(IAX2/carey)Nothing else? Try adding a Hangup command on the next priority and see if that helps any.exten = s,2,HangupIf you already have a Hangup command in there, then I apologize for wasting your time. :)Undrhil--- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: I
Re: [Asterisk-Users] asterisk-1.2.9.1
i guess you were right. it was due to the previous version of asterisk on my PC,although i had make clean it anyway thanx for the help. can you tell me if i can use the same iax.conf and extensions.conf files that i used for asterisk-1.0.3 for this 1.2.9.1 version? thanx again On 6/11/06, Thomas Kenyon [EMAIL PROTECTED] wrote: amna saleem wrote: hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error [pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! can anyone help me i have redhat linux enterprise zaptel version 1.2.6 libpri version 1.2.3 what am i missing here?I know it sounds daft, but could it be a module that was compiled aspart of a previous install of asterisk and wasn't overwritten/deletedand is being autoloaded (or directly loaded)? It's not a standard module (afaik) anymore, and certainly hasn't beencompiled with the versions I'm running.If it's for a particular card, you may need to recompile the moduleyourself from their driver source. If you don't know what it's used for, I'd move it out of/usr/lib/asterisk/modules, and see what happens/doesn't happen.Usually when you update asterisk, at the end of the make-install it willgive you a list of modules that are in the modules directory that it hasn't placed there, if it does that they are usually worth looking at.___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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] asterisk-1.2.9.1
Hi Amna, Can use all the archives * conf. In this case you will be making one upgrade of version of asterisk. I wait to have helped. Best RegardsJosué 2006/6/12, amna saleem [EMAIL PROTECTED]: i guess you were right. it was due to the previous version of asterisk on my PC,although i had make clean it anyway thanx for the help. can you tell me if i can use the same iax.conf and extensions.conf files that i used for asterisk-1.0.3 for this 1.2.9.1 version? thanx again On 6/11/06, Thomas Kenyon [EMAIL PROTECTED] wrote: amna saleem wrote: hi ! i have installed asterisk-1.2.9.1 but am unable to run it i am getting this error [pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Jun 11 16:43:00 WARNING[8968]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! can anyone help me i have redhat linux enterprise zaptel version 1.2.6 libpri version 1.2.3 what am i missing here?I know it sounds daft, but could it be a module that was compiled aspart of a previous install of asterisk and wasn't overwritten/deletedand is being autoloaded (or directly loaded)? It's not a standard module (afaik) anymore, and certainly hasn't beencompiled with the versions I'm running.If it's for a particular card, you may need to recompile the moduleyourself from their driver source. If you don't know what it's used for, I'd move it out of/usr/lib/asterisk/modules, and see what happens/doesn't happen.Usually when you update asterisk, at the end of the make-install it willgive you a list of modules that are in the modules directory that it hasn't placed there, if it does that they are usually worth looking at.___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] FXO registration and VegaStream
Hi Issac, Ok, here goes :) Again, my disclaimer-- I'm pretty new to Asterisk, so I'm sure half of this is not needed or potentially even misconfigured. You will even see some lines commented out, since I wanted to test if they were needed--they weren't. I'm hoping to clean everything up and put it on the wiki -- hopefully next week or two. Also, these are from Asterisk @ Home, so there might be some changes needed for your setup. * Sip.conf - context line may differ from [EMAIL PROTECTED] Defaults * [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf * * Sip_additional.conf - * I haven't tested DTMF on incoming calls-- you may have to * change dtmfmode to inband (rfc2833 didn't work for the outgoing * calls). Also, the context may need to be changed for security? * I only have an entry for 01 since I am testing with 1 line only * ... snip ... [01] ;most lines added by [EMAIL PROTECTED], may not be necessary (i.e. mailbox) username=01 type=friend secret=...my vega's password for line 1... (see POTS in Vega's web config) record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 01 ... snip ... ; commented out, doesn't seem to be needed ;[vega] ;type=user ;dtmfmode=inband ;disallow=all ;context=from-pstn ;allow=ulaw [vega-gw] type=peer host=192.168.1.30 ; my vega's IP address dtmfmode=inband ;DTMF doesn't work with rfc2833, unfortunately disallow=all ;context=from-internal ; commenting out, makes context default to from-sip-external? allow=ulaw ;only allow ulaw * * extensions_additional.conf - dials extension 106 on incoming * call. I think there's some special [EMAIL PROTECTED] magic happening in the * macro to dial 106. You could just have something like Dial() * happen here. * * After adding the 06 extension, that is when incoming calls * start going through. * * You could also use the s extension somehow, as Mike showed us * (I need to read up a little!! :) ) * exten = 06,1,Macro(exten-vm,novm,06) exten = 06,hint,SIP/106 * * Configuration Change Report from the Vegastream * (shows changes from factory settings) * Report on configuration changes (verbose) Configuration changes: Key: CU: Changed from factory and unsaved. C-: Changed from factory and saved. -U: Not changed but unsaved. [call_control.timers.1] T301_timeout=90 T301_cause=18 [dsp.g711Alaw64k] VADU_threshold=0 VP_FIFO_max_delay=160 VP_FIFO_nom_delay=60 echo_tail_size=16 idle_noise_level=-7000 packet_time_max=30 packet_time_min=10 packet_time_step=10 rx_gain=0 tx_gain=0 [dsp.g711Alaw64k.data] EC_enable=disable [dsp.g711Alaw64k.voice] EC_enable=enable [dsp.g711Ulaw64k] ;I'm only using Ulaw, so this is the only codec set up VADU_threshold=0 C- VP_FIFO_max_delay=60 *factory=160 C- VP_FIFO_nom_delay=10 ; I figured reducing this is ok (Asterisk - vega is on a LAN), and might reduce delay? *factory=40 C- echo_tail_size=8 ; EC trains much faster @ 8ms tail for me (we are close to CO) *factory=16 idle_noise_level=-7000 C- packet_time_max=20 ; Asterisk requires 20ms packets for ULAW *factory=30 C- packet_time_min=20 ; Asterisk requires 20ms packets for ULAW *factory=10 packet_time_step=10 rx_gain=0 tx_gain=0 [dsp.g711Ulaw64k.data] EC_enable=disable [dsp.g711Ulaw64k.voice] EC_enable=enable [dsp.g729AnnexA] VADU_threshold=0 VP_FIFO_max_delay=500 VP_FIFO_nom_delay=60 echo_tail_size=16 idle_noise_level=-7000 packet_time_max=80 packet_time_min=10 packet_time_step=10 rx_gain=0 tx_gain=0 [dsp.g729AnnexA.voice] EC_enable=enable [dsp.g729] VADU_threshold=0 VP_FIFO_max_delay=500 VP_FIFO_nom_delay=80 echo_tail_size=16 idle_noise_level=-7000 packet_time_max=80 packet_time_min=10 packet_time_step=10 rx_gain=0 tx_gain=0 [dsp.g729.voice] EC_enable=enable [dsp.g7231] VADU_threshold=0 VP_FIFO_max_delay=500 VP_FIFO_nom_delay=30 echo_tail_size=16 idle_noise_level=-7000
Re: [Asterisk-Users] JIAX status
Rubens Zupelli Filho [EMAIL PROTECTED] writes: You are compiling in Linux or Windows? Both. It works on Linux, but not yet on Windows. The package the java compiler is not founding is: net.sourceforge.iaxclient.jni That's part of the source package; probably the classpath just needs to be tweaked. ---Scott. ___ --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: Nokai E60 and E61 , working fine with Asterisk , with new access points
Hi, Im an unsuccessful user of E60. Please post the configs on the phone in detail Thanks Dan On 11/06/06, Markus Schuster [EMAIL PROTECTED] wrote: John Joseph wrote: Was able to communicate clearly with e60 and E61 with asterisk with new access point [..] Could you please post some details (or even better: write them in some sort of Wiki) on the configuration you did on the Nokia? I'm thinking about buying a Nokia E60 but after a short web search there seem to be some problems about the correct configuration of the phone. Greetings, Markus ___ --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] quad t1 / 1U rack server combos
Hy men, :-) Use Industrial PICMG PC's. Higher cost at buy, but very stable and evolutive platforms. SBC doesn't change during a long industrial period. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steve Totaro Envoyé : lundi 12 juin 2006 01:06 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] quad t1 / 1U rack server combos Colin Anderson wrote: C'mon guys! Certify a few current model servers and be done with it. Problem is, certification is a moving target and can become invalid with something as simple as a BIOS change by the manufacturer. Now that the barrier to entry to changing a design is almost nil, manufacturers love to screw around with designs to save a few bucks. I have seen two identical boxes, labelled as such by the manufacturer, bought in the same time period, but with different guts. Digium would wind up with egg on their faces by certifying a system, then 90 days later after everyone buys it, finds out that some subtle change by the manufacturer has destabilized the config. I agree it is frustrating as hell, but this is the price we pay. Would you rather buy a Mitel for 10X the $$$? Maybe in some circumstances, it is worth it. -- -- lman/listinfo/asterisk-users I bought two HP DL380s at the exact same time from CDW. I used the first one to build an image to transfer to the other system. When I booted the second system, kudzu reported different NICs. So, yes, next to impossible to certify any hardware without the hardware manufacturer's help. ___ --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] What does RELAXDTMF do?
Thanx for everyone's passionate responses, and apologises for not replying sooner. 1. Based on what I have seen I take it noone is sure of what the true purpose and the effects of the relaxdtmf parameter offer. 2. I am using both a mixture of VSP's and SPA3K's, but primarily it is the VSP's 3. And as for tweaking the output level's, I see no options being provided within the IAX and SIP configuration files, and not really concerned with the SPA3K's at this time since they are fail-overs. 4. Since the bulk of my issues are through the IAX2/SIP methods for DMTF and Asterisk has direct issues with DTMF, I just hope that v1.4 will be released sooner rather than later. On 09/06/2006, at 5:11 AM, Martin Joseph wrote: On Jun 8, 2006, at 3:13 AM, Peter J Dean wrote: I have an issue with DTMF. DTMF is being partly recognised by some external IVR systems (banks, billing, etc), other IVR systems have intermittent issues. Call our VSP directly and using their IVR system without issue, and our internal IVR works just fine. Currently i have all voip devices using RFC2833, which is what is recommended, and thus the sip.conf file has dtmfmode=rfc2833 and relaxdtmf=yes. I have not seen any information that clearly defines the purpose of the relaxdtmf parameter in the sip.conf file, and wondering of flicking it from yes to no will have an impact, and if so what sort of impact will it have? Dunno about that, but usually the gain of the output is the most likely source for this kind of intermittent issue. If you can tweak your output level a hair each way and retest, you might find you are ok. Just a thought, marty ___ --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
