Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sun, 2004-12-19 at 02:11, David Uzzell wrote: Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! Sorry, but maybe you should have read his posts more thoroughly. ztdummy is not an option because of his chipset. He has usb-ohci. ztdummy requires usb-uhci. Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Bruno Hertz wrote: On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote: I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. The problems you had and were resolved with ztdummy, were they primarily IAX related ? Since, after all, the main channels relying on special timers are Meetme, IAX and (maybe) MusicOnHold according to http://www.voip-info.org/wiki-Asterisk+timer Just want to be sure, since I still believe my mere demo playback issue likely has a different reason ... I'd like to chime in here as I have a similar problem. I have been toying with * on other (cheapo) hardware not so successfully (mainly due to the audio chipsets). I just purchased an ASUS AV8 (Socket 939 Athlon 64 3500+) system for my real world testing, it's a high end MB and overall it has 98% of the feature set for what I wanted to accomplish. Currently I'm running FreeBSD 5.3 under the amd64 port of the OS (fyi). I'm experiencing the exact same symptoms - choppy clicking of the demo voice. I'll start by saying that I have done a reasonable amount of research on *, MB chipsets, and FreeBSD, and I've spent considerable time getting the basic functionality to work. The ports version of * under FreeBSD needed some tweaking to work under amd64 vs i386, but I have a working version including h323 and oss that works with the demo stuff. From what I have read the issue with choppy sound under the demo voice seems to be due to a timing issue, one that can't be solved under FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as far as USB stuff that will handle this. I do not have a Digium card installed yet, but I will have a TDM400P in a couple of days. Will a Digium card with the current driver solve the problem ? (zaptel doesn't compile for FreeBSD 5.3 amd64, maybe for i386). Given that I have a working installation with the same symptoms as reported, I'm leaning towards us having the same problem. If this is a timing issue, it would be great to solve this in a systematic way (without external hardware). Thoughts? Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free World Dialup and Asterisk
Yes, of course you can do that. I have this very setup working for the office, with * aggregating voip and isdn incoming calls and forwarding them to my laptop wherever I am. just follow the instructions on the FWD website, and run iax2 debug from the console to see what's happening in anything goes wrong. l. In data Sat, 18 Dec 2004 20:33:01 -0800 (PST), Gonzalo Gasca Meza [EMAIL PROTECTED] ha scritto: Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD dialup account - Internet -- 3COM router/switch --- Asterisk -- 7960 -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE or IAX?
On Sun, 19 Dec 2004, Eric Bishop wrote: Apart from the the coolness factor can anyone explain to me in what situation one would use TDMoE rather than IAX for communication betwwen 2 Asterisk servers? There are two advantages with TDMoE: * low latency (prevents far end echo from going from nice sidetone to irritating percevied echo) * supports full pri signalling (hangupcause, type of number etc) There are disadvantages as well compared to iax: * non routeable (local ethernet only) * channels have to be preconfigured * more? I guess the key factor is if you need the low almost-tdm latency. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Martin List-Petersen wrote: On Sun, 2004-12-19 at 02:11, David Uzzell wrote: Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! Sorry, but maybe you should have read his posts more thoroughly. ztdummy is not an option because of his chipset. He has usb-ohci. ztdummy requires usb-uhci. Umm yes it does on 2.4 kernel but on a 2.6 kernel it doesn't cause I am running it on a 2.6 kernel and I don't have that hardware. Quoted from http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer On kernel version 2.6 it uses internal high-resolution kernel timer and do not require any additional hardware. Now in the original post he says that he is using FC2 so I am not 100% sure if it is 2.6 or 2.4 but FC2 is only one step away from FC3 which does run a 2.6 kernel. I don't know on FC2 as I have never run it. And yes to answer the original poster it did solve my IAX problems. With the demo I would sugest that maybe the SMP kernel on a single CPU server could be a partial cause. I have seen strange things on Dual CPU servers running SMP kernels were 1 CPU has been removed. Hope that helps. David Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call screening
Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call screening
Yes U can do it with asterisk and by dialing *98 on your Ip Phone you can listen to your message -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Sunday, December 19, 2004 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call screening Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call screening
Sorry, I don't follow. Dialing *98 will achieve what? Up until the time I decide to take the call, I want to be able to hear the person leaving a message interactively, so when I decide to pick up the call he's still there. Like a regular answering machine -Original Message- From: hadi [mailto:[EMAIL PROTECTED] Sent: Sunday, December 19, 2004 12:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] call screening Yes U can do it with asterisk and by dialing *98 on your Ip Phone you can listen to your message -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Sunday, December 19, 2004 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call screening Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN HFC cards
Hi, Currently I am using a ISDN BRI PCI FRITZ card (works), would I get any benefits switching to a HFC card? Or it would be a better choice to switch to a ISDN with a DSP processor? Currently I have echo on my CAPI channel when calling analog lines, if call a cell phone, ISDN or a PRI PBX it doesn't show up any echo. So this indicates a far-end echo, how can this be minimized? I turned on the Squelch on the capi and it works but during a conversation the sometimes I tend to get small click and it distracts a little bit, even tough it is still better than the echo. If switching to HFC works better can someone point out where to buy them (online)? Regards, Humberto Aicardi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call screening
Sorry I mean the voice mail -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Sunday, December 19, 2004 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] call screening Sorry, I don't follow. Dialing *98 will achieve what? Up until the time I decide to take the call, I want to be able to hear the person leaving a message interactively, so when I decide to pick up the call he's still there. Like a regular answering machine -Original Message- From: hadi [mailto:[EMAIL PROTECTED] Sent: Sunday, December 19, 2004 12:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] call screening Yes U can do it with asterisk and by dialing *98 on your Ip Phone you can listen to your message -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer Sent: Sunday, December 19, 2004 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call screening Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Per extension/user CDR?
It seems that all my CDR is dumping into the Master.csv file. There is a way to create per user/extension CDR but I have looked endlessly in the Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find a way to do this.. Any help would be appreciated. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call waiting/ 3 way calling
HI; I have an Asterisk with 10 "SIP" ip-phones, our pbx features are now: Voicemail and Call Transfer. How can I serve both "Call Waiting / 3 way calling" for our SIP Phones.?/ Appreciate Any Help Mohammad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. The problems you had and were resolved with ztdummy, were they primarily IAX related ? Since, after all, the main channels relying on special timers are Meetme, IAX and (maybe) MusicOnHold according to http://www.voip-info.org/wiki-Asterisk+timer Just want to be sure, since I still believe my mere demo playback issue likely has a different reason ... I'm 95% sure iax is not dependent on the ztdummy type timers. Maybe the OP could give us a little more detail on the specific data flow that he's having an issue with. I interpreted his call problem as: sipdev1 - ? - teliax.com - iax - OP-asterisk - sipdev2 He indicated sipdev1 was running VAD, and the call was completed via teliax.com to his asterisk with crackly audio. If this is the case, the issue is VAD between sipdev1 and the ? box shown in the data flow. Since there isn't a consistent flow of rtp data packets between sipdev1 and ? because of VAD, what gets sent to teliax.com is already choppy audio. There is nothing the OP is going to be able to fix between teliax.com and sipdev2 to correct for a problem that is located elsewhere. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phone choices....opinion request Polycom vs Cisco
Hi I am struggling with hardware choices to get started with. My options are narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. of importance is: - functionality / integration with asterisk - headset functionality and use - voice quality - build quality Is there much of a difference between Polycom and Cisco? Scanning the group it looks like there may be slightly more issues with Polycom but I don't know how they stack up on the integration with Asterisk and future flexability. Any recommendations appreciated. Thanks Walt _ Don't just search. Find. Check out the new MSN Search! http://search.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
http://www.voip-info.org/wiki-RTP+Silence+Suppression http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html So I am confused. The first says that VAD is supported in RTP. Ok, I know that. The second is kinda ambiguous and seems to imply that * doesnt support VAD. I think it does now as there is a VAD=yes option in SIP.conf. Either way since IAX doesnt use RTP both statements are probably not relevant. Does * support VAD with IAX? If so can it be turned on and off in IAX?? Does anyone know definitively?? I really like to turn it off and just send packet continuously. Should I file a bug (feature request)?? Looking at the current /usr/src/asterisk/configs/sip.conf.sample, VAD=yes does not exist. Since those sample files tend to be the formal documentation for valid asterisk parameters, it should be safe to say its not supported. Same for iax.conf.sample; doesn't exist there either. The comment made by John Todd in the August 2003 posting was simply suggesting to the original poster (as that time) that he should enter a feature request into the asterisk bug tracker if he felt strongly that VAD was needed. The description of VAD in the voip-info reference is simply someone documenting what the sip rfc states about VAD. It does not imply or even suggest that asterisk supports VAD. Asterisk does not support VAD today (nor does it support every option documented in the sip rfc). The iax data flow betwen two boxes is not the same as sip-rtp data flows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call billing
how do youintegrateGnugk and Asterisk billing? Are you using Asterisk's H323 channel?Voip Business [EMAIL PROTECTED] wrote: I integrate Gnugk and a gnugk billing system working like a charm.regardsHAOn Sat, 18 Dec 2004 01:48:56 -0800, Inam <[EMAIL PROTECTED]>wrote: HI Alll this is my first post on users list can any body let me know how can one integrate his/her billing applications to Asterisk Softswitch Thanks in advance INAMULLAH KHOSA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemailmain hotkey
On Sun, Dec 19, 2004 at 12:21:28AM -0600, Matthew Boehm wrote: I'm having a similar problem. Do you have operator=yes in your voicemail.conf under [general]? Argh, thats it, solved! Thanks a lot :) ...cut -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Make asterisk launch script after completing call.
OK. I now have call recording working for both incoming and outgoing
calls.
Now I want to make those wavs into mp3. I could launch a script from
cron that checks for new wavs and converts them. But that wouldn't be
so elegant.
Launching it from * on hangup would be nicer. How is it done?
[outgoing]
exten = _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _0.,2,Monitor(wav,${CALLFILENAME},m)
exten = _0.,3,Dial(SIP/rix/${EXTEN}|20|t)
exten = _0.,4,Congestion
exten = _0.,104,Congestion
[sip-in]
exten = 1000,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 1000,2,Monitor(wav,${CALLFILENAME},m)
exten = 1000,3,Dial(SIP/alex,20)
exten = 1000,4,Voicemail(u1000)
--
Alex Polite
http://polite.se
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Re: [Asterisk-Users] Free World Dialup and Asterisk
He's trying to use sip, not iax. It would appear he's got both a fwd registration issue and an incoming fwd context issue. They don't appear to be in sync (probably an understanding of context issue actually). Yes, of course you can do that. I have this very setup working for the office, with * aggregating voip and isdn incoming calls and forwarding them to my laptop wherever I am. just follow the instructions on the FWD website, and run iax2 debug from the console to see what's happening in anything goes wrong. l. In data Sat, 18 Dec 2004 20:33:01 -0800 (PST), Gonzalo Gasca Meza [EMAIL PROTECTED] ha scritto: Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD dialup account - Internet -- 3COM router/switch --- Asterisk -- 7960 -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Siemens HiCom PBX with Asterisk through E1
Hi I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4 and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard takes a few seconds and sets the link to green (OK). 2. I've tried to connect our running E1 line from the telco with wildcard. The modem (modulates HDSL to G703 120 Ohms). I've used a 1:1 cable that did not work. I even tried to connect the copper wires by hand which resulted that the modem gave me a green power light but Wildcard stayed on a waving red light. 3. I have plugged out our running PBX and connected it to Wildcard which resulted in a green light for one second and then the state from zttool switched to yellow (and Wildcard to constant red light). The protocol used by the modem is hdb3 and /etc/zaptel.conf is adjusted according to this. Can anyone clarify the different protocol layers and when fails what? When occurs the green light? Must protocol layer2 be established or even higher or is it just a layer1 link light. Please help me out. Jens ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free World Dialup and Asterisk
Gonzalo,
Have you tried IAX, I see yo are behind NAT, and my experiences with IAX
behind NAT are much less painful :-)
I've FWD via IAX, receiveing calls (in fact, last time was a nearby
person in Portugal :-) that tested it).
One last thing, you mention dialup client, I guess she is not in dialup,
correct? From what I recall, FWD would do only G.711, would not exactly
work in dialup (maybe ISDN with 2 b-channels ?)
PS: I don't see the dialplan for the inbound calls, where a call from
FWD would land in your * ?
Gonzalo Gasca Meza wrote:
Hi forum,
I have been fighting days and days configuring FWD and asterisk with NO success
I have the following scenario.
My sister in Spain with FWD dialup client
My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone.
Spain LAN
FWD dialup account - Internet -- 3COM router/switch --- Asterisk -- 7960
I have done some research in google with no success.
http://www.m-networks.net/home/asterisk/ast-fwd.htm
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
When I connect my FWD client in the LAN i can dial FWD numbers
ANY IDEAS OR CONF FILES WORKING WILL BE APPRECIATED
THANKS!
server*CLI sip show registry
Host Username Refresh State
69.90.155.70:5060 431044 160 Registered
69.90.155.70:5060 421058 160 Registered
SIP.conf
register = 421058:[EMAIL PROTECTED]/103 ;Register Free World Dialup
register = 431044:[EMAIL PROTECTED]/103
[fwd1]
type=friend
username=431044
secret=password
fromuser=431044
fromdomain=fwd.pulver.com
host=fwd.pulver.com
insecure=very
canrenvite=no
nat = yes
dtmfmode=inband
[fwd2]
type=friend
secret=password
username=421058
fromuser=421058
fromdomain=fwd.pulver.com
host=fwd.pulver.com
dtmfmode=inband
nat=yes
canreinvite=no
extensions.conf
FWDUSERID1=421058
FWD1USERNAME=Gonzalo Gasca
FWDUSERID2=431044
FWD2USERNAME=Gonzalo Gasca
FWDPREFIX=*
[fwd1-out]
exten = _8.,1,SetCallerID(${FWDUSERID2})
exten = _8.,2,SetCIDName(${FWD2USERNAME})
exten = _8.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten = _8.,4,Macro(fastbusy)
exten = _8.,5,Hangup
[fwd2-out]
exten = _7.,1,SetCallerID(${FWDUSERID1})
exten = _7.,2,SetCIDName(${FWD1USERNAME})
exten = _7.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],70)
exten = _7.,4,Macro(fastbusy)
exten = _7.,5,Hangup
My IP phone include those fwd1-fwd2-out
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[Asterisk-Users] Re: Call Screening
Hello-- I've done some coding for call screening in Asterisk. It's not in Asterisk yet, mainly because we're waiting for prompts from Allyson so it sounds like the rest of the system. But patches, prototype sound files, etc, are all filed at: http://bugs.digium.com/bug_view_page.php?bug_id=752 And I'd love to have your feedback. murf Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Termination
On Sat, Dec 18, 2004 at 06:28:54PM +, Antony Stone wrote: On Saturday 18 December 2004 18:07, Dorn Hetzel wrote: I wouldn't say I hate SIP, it sucks less than H.323 and so on by a large margin. But, having said that, if you can use IAX, it sucks even much than SIP does :) Um, are you saying IAX is good, or that it is not good? I'm not sure I understand your statement above. If you are saying that IAX is bad, why? And what's better? EEK!!! Darn rented fingers :) s/even much than/even much less than/ For my money, IAX is the best solution if you can use it, or put another way, It sucks the least of all available options :) -Dorn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BRI Error with zaphfc
From the trace it appears that you are not getting any Layer 2 communication. All the broadcast messages like SETUP and the TEI assignments are being sent (and because your phone rings, it is hearing what Asterisk is saying to it. Your ISDN phone does not appear to be responding. This looks to me like a physical connection problem. Try checking the terminations - you MUST have 100 ohm resistors across both TX and RX pairs. In fact, on short cable runs you are sometimes better off with 50 ohms to simulate the 100 ohm terminators expected at each end of the bus. Rgds Tim Ian Clough wrote: Hi I have a small ISDN PABX , (BRI) at home - Siemens Gigaset 4175 which has cordless DECT extensions. I have set up asterisk on FC3 with two HFC cards and I am using the latest bristuff. I am trying to use * between my ISDN line and my PABX. One card is in TE mode and can receive and make calls OK. I can make and receive external calls to a SIP phone. I cannot get the other card to talk to my PABX. It is in NT mode. If I try to call a PABX extension it rings but drops the call after a few seconds. The PABX logs the call as a 'missed call' I get the following debug:- I understand Linux better than ISDN. Can anybody interpret this for me please :-) Ian pri intense debug span 2 Enabled EXTENSIVE debugging on span 2 -- Accepting call from '12' to '--' on channel 0/1, span 1 (my tel no deleted :-) -- Executing Dial(Zap/1-1, Zap/g2/673615) in new stack [ 02 ff 03 08 01 02 05 04 03 80 90 a3 18 01 89 70 07 c1 36 37 33 36 31 35 a1 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 22 bytes of data Protocol Discriminator: Q.931 (8) len=22 Call Ref: len= 1 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 89] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] [70 07 c1 36 37 33 36 31 35] Called Number (len= 9) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '673615' ] [a1] Sending Complete (len= 1) -- Called g2/673615 [ fc ff 03 0f 59 76 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI assign ri=22902 tei=64 [ fe ff 03 0f 59 76 02 81 ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data [ fc ff 03 0f 02 8f 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI assign ri=655 tei=65 [ fe ff 03 0f 02 8f 02 83 ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data [ fc ff 03 0f 26 d7 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI assign ri=9943 tei=66 [ fe ff 03 0f 26 d7 02 85 ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data [ fc ff 03 0f 05 06 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI assign ri=1286 tei=67 [ fe ff 03 0f 05 06 02 87 ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data -- Channel 0/1, span 2 got hangup Dec 19 08:21:47 WARNING[-188937296]: app_dial.c:406 wait_for_answer: Unable to forward voice -- Hungup 'Zap/4-1' == No one is available to answer at this time -- Executing Wait(Zap/1-1, 14) in new stack [ fc ff 03 0f 35 eb 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI assign ri=13803 tei=68 [ fe ff 03 0f 35 eb 02 89 ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data [ fc ff 03 0f 45 c9 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI assign ri=17865 tei=69 [ fe ff 03 0f 45 c9 02 8b ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data --
Re: [Asterisk-Users] 3rd party call control / CSTA , JTAPI or TAPI interfaces
On Fri, 2004-12-17 at 16:47 -0800, Shahed wrote: Hello all, (Not sure if this is more appropriate for user or dev list) Does asterisk have any sort of standards based api that can enable an application to do call control on the switch ? For example, if I am developing a call center application using asterisk, I would like to be notified of inbound calls and then be able to route them to extensions / agents based on my application logic. Maybe you need to learn about how flexible the dialplan is. Extensions.conf is pretty close to being a full featured programming language of it's own. Also, don't cross-post. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
extensions.conf:
exten = _8500, 1, Wait(2)
exten = _8500, 2, VoicemailMain(${CALLERIDNUM})
exten = _8500, 3, Hangup
You don't mention the type of phone you're using, but on our setup with
SIP phones, we add a sipdtmfmode(inband) to what you have above. You
might try fiddling with that.
-russ
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Re: [Asterisk-Users] Grandstream CallerID
On Sunday 19 December 2004 06:31, Wilson Pickett wrote: Is it possible to send the incoming PSTN caller ID to a Grandstream Budge Tone-100 SIP phone? I've configured the extensions.conf file and the log is As Eric notes, the BT100 phones won't show letters. If a call comes in without CID, asterisk sends a string like Asterisk call which the BT will try to display as some giberish so I have setcallerid to 000 when this happens. I'd recommend using setcallerid(1234567890) (or any number) to test the phone, which should display that. If callerid does come in from PSTN, it should just make through as he said. I'm having the same issue as David, and forcing the CallerID with SetCallerID() doesn't work - it still only shows the extension of the phone I'm calling. (Yes, I've checked the CDR and also done a show channel on the phone while ringing - the Caller ID is reset properly I think I have the solution -- I had the fromuser= variable set, which is what Asterisk uses to force CallerID as when it's making calls out to the Grandstream. Is that set, David? If so, unset it, and you should be fine. --mec ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
It BT100. it works.
thanks!
steven
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Russ
Beaupre, P.E.
Sent: Sunday, December 19, 2004 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoicemailMain can't read from phone
keyboard!
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called,
it
prompts for mailbox and password. It seems not able to read from the
phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
extensions.conf:
exten = _8500, 1, Wait(2)
exten = _8500, 2, VoicemailMain(${CALLERIDNUM})
exten = _8500, 3, Hangup
You don't mention the type of phone you're using, but on our setup with
SIP phones, we add a sipdtmfmode(inband) to what you have above. You
might try fiddling with that.
-russ
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Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On Sun, 2004-12-19 at 00:40 -0800, Chris Miller wrote: From what I have read the issue with choppy sound under the demo voice seems to be due to a timing issue Taking the risk of appearing notorious, I again emphasize that I don't believe that. I have asterisk right now with ztdummy running on a Debian Sarge box. When I connect with either GnomeMeeting/oh323 or iaxcomm via home LAN to that box, I experience exactly that symptoms. i.e. choppy demo voice. Now, if I boot FC3 on that same box, with the same asterisk version compiled under FC3, I *do not* get choppy sound even without ztdummy. Actually, I never bothered compiling zaptel support on FC3. Of course, I hope the card you're expecting will solve that problem for you, but I wouldn't be suprised if it didn't. Regards, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM120 card?
[EMAIL PROTECTED] wrote: hi any chance of making asterisk support these? http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-3835624908 8.htm According to the manufacturer, they already do: http://www.ipvolution.com/ Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialplan selection
Hello, I would like to parse inbound Asterisk IAX2 7-digit numbers in the form of 123-4567 and strip out the first four digits, and then dial whatever number digits remain. If I only have three digits (000-999) and have a mix of channels (ZAP, SIP, IAX2) could someone please point out how I can use a single DIAL command to just dial the extension regardless of the type of channel. .. For each valid extension, I have a separate dial command anyway, which denotes the particular channel that extension is assigned to. I do not want to assign groups of extensions i.e., 123-A567 or 123-B567 or 123-C567 where A=ZAP, B=SIP, C=IAX2 peers respectively. -samudra ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting/ 3 way calling
mohammad wrote: I have an Asterisk with 10 SIP ip-phones, our pbx features are now: Voicemail and Call Transfer. How can I serve both Call Waiting / 3 way calling for our SIP Phones.?/ This is what I call one of the dirty little secrets of SIP. On SIP phones (and H323) all the call control is done by the PHONE itself, not by the PBX. Some SIP phones do not even support 3-way calling or supervised transfers (the BT101 comes to mind). There really isn't anything Asterisk can do to make it work if the phone does not support the feature. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting the real extension into CDR
Matthew Boehm wrote: Hey gang, Getting ready to run some test bills for customers. Most SIP phones have both an extension and a DID. If a person calls a DID asterisk redirects the call to the right extension: exten = 8005551212,1,Goto(companyA-internal,3022,1) The problem is, that if someone calls 8005551212, the CDR shows the DST number as 3022. Is there a way around this? I understand that 3022 is the destination but that isn't what the outside person dialed; and I need to know what was dialed in order to bill correctly. The same goes for the 's' extension. Lots of my CDRs show 's' as the DST instead of the actual number the person dialed. I see there are a few apps for modifying the CDR, but I don't see anything to let me modify the DST number. The only way I know of to do what you want to do is to not use the GoTo, but to set up extension 8005551212 exactly like extension 3022. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
Steven Wang wrote: Hello I try to set up voicemails for extension. When VoicemailMain gets called, it prompts for mailbox and password. It seems not able to read from the phone. So the authentication always fails. This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch between the phone and Asterisk. For most phones you want to use RFC2833 for both the phone and for the entry for that phone in sip.conf. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM120 card?
Jim Van Meggelen wrote: According to the manufacturer, they already do: http://www.ipvolution.com/ Wow... if that board actually ships as promised, with Asterisk support, that will be amazing. Up to 8 T1/E1 in a singe PCI slot, with onboard codecs and echo cancellation... and a price that is very competitive. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On 19/12/2004 16:40 Chris Miller said the following: seems to be due to a timing issue, one that can't be solved under FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as the ztdummy pseudo timer works well under freebsd 4.x and 5.x. i used it for a bit before i got my digium cards. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
On 19/12/2004 20:38 Rich Adamson said the following: I'm 95% sure iax is not dependent on the ztdummy type timers. trunked iax requires a timer, either ztdummy or a digium card. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream CallerID
I did have fromuser set in my sip.conf so I went in and commented the line
out (thanks for the help on that). This is what I have in my
extensions.conf file:
exten = s,1,SetCallerID(${CALLERID}) ; Set the caller ID
exten = s,2,Wait(2)
exten = s,3,Dial(SIP/1234,20,tr) ; Dial our office SIP phone
exten = s,4,Hangup
That produces the full CID coming from the PSTN (Verizon) in the log
something like:
Joe Somebody 7035551212
When I change the SetCallerID() function to:
exten = s,1,SetCallerID(111) ; Set the caller ID
It still doesn't send the CID to the phone (or if it does, the phone isn't
recognizing it). Do I have my extensions.conf file setup wrong? Is there
something special I need to do in the SIP phone configuration? Here's the
SIP section of the /etc/asterisk/sip.conf:
[1234]
type=friend
context=sip
username=1234
secret=
callerid=Office 1234
host=dynamic
nat=no
canreinvite=yes
dtmfmode=inband
incominglimit=1
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
I noticed on the WiKi there's a function called SetCIDNum(), but that
doesn't do anything (or at least nothing I could see). I'm stumped on this
one. Any advice is always welcomed. :)
Thanks,
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Clauson
Sent: Sunday, December 19, 2004 11:44 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
On Sunday 19 December 2004 06:31, Wilson Pickett wrote:
Is it possible to send the incoming PSTN caller ID to a Grandstream
Budge Tone-100 SIP phone? I've configured the extensions.conf file
and the log is
As Eric notes, the BT100 phones won't show letters. If a call comes
in without CID, asterisk sends a string like Asterisk call which
the BT will try to display as some giberish so I have setcallerid to
000 when this happens. I'd recommend using setcallerid(1234567890)
(or any number) to test the phone, which should display that. If
callerid does come in from PSTN, it should just make through as he
said.
I'm having the same issue as David, and forcing the CallerID with
SetCallerID() doesn't work - it still only shows the extension of the
phone I'm calling. (Yes, I've checked the CDR and also done a show
channel on the phone while ringing - the Caller ID is reset properly
I think I have the solution -- I had the fromuser= variable set, which
is what Asterisk uses to force CallerID as when it's making calls out
to the Grandstream. Is that set, David? If so, unset it, and you
should be fine.
--mec
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Re: [Asterisk-Users] Connecting Siemens HiCom PBX with Asterisk through E1
On Sun, 19 Dec 2004, Jens Kübler wrote: I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4 and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard takes a few seconds and sets the link to green (OK). It should be 1 to 4 and 2 to 5, not 3. The pairs are 1-2 and 4-5. 2. I've tried to connect our running E1 line from the telco with wildcard. The modem (modulates HDSL to G703 120 Ohms). I've used a 1:1 cable that did not work. I even tried to connect the copper wires by hand which resulted that the modem gave me a green power light but Wildcard stayed on a waving red light. That should have worked, as long as 1-1, 2-2, 4-4 and 5-5. Are you sure the signalling is correct? What did zttool say? 3. I have plugged out our running PBX and connected it to Wildcard which resulted in a green light for one second and then the state from zttool switched to yellow (and Wildcard to constant red light). Yellow alert is remote alert, right? That would indicate that the path from the pbx to asterisk is ok, but not the path from asterisk to the pbx. The protocol used by the modem is hdb3 and /etc/zaptel.conf is adjusted according to this. Do you have crc4 enabled? Care to post your zaptel.conf. I think you have to start asterisk on the span to get the upper layers enabled. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream CallerID
I forgot to ask, since the BT100 can't take characters (only numbers), I
would have assumed that there was a function to extract a number from an
incoming PSTN CID, is that possible?
Thanks again,
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ishmael
Sent: Sunday, December 19, 2004 2:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Grandstream CallerID
I did have fromuser set in my sip.conf so I went in and commented the line
out (thanks for the help on that). This is what I have in my
extensions.conf file:
exten = s,1,SetCallerID(${CALLERID}) ; Set the caller ID
exten = s,2,Wait(2)
exten = s,3,Dial(SIP/1234,20,tr) ; Dial our office SIP phone
exten = s,4,Hangup
That produces the full CID coming from the PSTN (Verizon) in the log
something like:
Joe Somebody 7035551212
When I change the SetCallerID() function to:
exten = s,1,SetCallerID(111) ; Set the caller ID
It still doesn't send the CID to the phone (or if it does, the phone isn't
recognizing it). Do I have my extensions.conf file setup wrong? Is there
something special I need to do in the SIP phone configuration? Here's the
SIP section of the /etc/asterisk/sip.conf:
[1234]
type=friend
context=sip
username=1234
secret=
callerid=Office 1234
host=dynamic
nat=no
canreinvite=yes
dtmfmode=inband
incominglimit=1
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
I noticed on the WiKi there's a function called SetCIDNum(), but that
doesn't do anything (or at least nothing I could see). I'm stumped on this
one. Any advice is always welcomed. :)
Thanks,
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Clauson
Sent: Sunday, December 19, 2004 11:44 AM
To: Wilson Pickett; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream CallerID
On Sunday 19 December 2004 06:31, Wilson Pickett wrote:
Is it possible to send the incoming PSTN caller ID to a Grandstream
Budge Tone-100 SIP phone? I've configured the extensions.conf file
and the log is
As Eric notes, the BT100 phones won't show letters. If a call comes
in without CID, asterisk sends a string like Asterisk call which
the BT will try to display as some giberish so I have setcallerid to
000 when this happens. I'd recommend using setcallerid(1234567890)
(or any number) to test the phone, which should display that. If
callerid does come in from PSTN, it should just make through as he
said.
I'm having the same issue as David, and forcing the CallerID with
SetCallerID() doesn't work - it still only shows the extension of the
phone I'm calling. (Yes, I've checked the CDR and also done a show
channel on the phone while ringing - the Caller ID is reset properly
I think I have the solution -- I had the fromuser= variable set, which
is what Asterisk uses to force CallerID as when it's making calls out
to the Grandstream. Is that set, David? If so, unset it, and you
should be fine.
--mec
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[Asterisk-Users] Looking for new hardware
Hi. I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm going to install: 1-)One X100P (1 FXO module) 2-)One TDM03B (3 FXO modules) I'll have the 4 FXO channels busy almost all the time, and I would like quality to be as good as possible without going to the high-level hardware. I would like to learn of some tested configurations (I've heard of problems with VIA chipsets, PCI voltages, etc). PC will be asterisk dedicated, so I'll like it to be asterisk+DigiumCards optimized. Can you recommend something? Thanks in advance, RODOLFO ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream CallerID
I forgot to ask, since the BT100 can't take characters (only numbers), I would have assumed that there was a function to extract a number from an incoming PSTN CID, is that possible? Try this exten = s,5,SetCIDNum(1234) and see if the phone displays it ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch between the phone and Asterisk. For most phones you want to use RFC2833 for both the phone and for the entry for that phone in sip.conf. Yep, and the BT will only work right with certain codecs. I think it's iLBC that suddenly won't recognize DTMF while it works with the same setting in ULAW, for example. I keep forgetting why I don't use iLBC on the BT, set it up, and then find DTMF b0rken with dtmfmode=info ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SMS - how to send one
I've read quite a bit in the older mailing list posts and the wiki but
I'm missing some simple point.
1) What is required to send an SMS to a mobile outside the office given:
Channel: ZAP/1
send it to $SMS_RECIPIENT (which includes the final extra digit)
via
$SMS_CENTER=the national message center server for sending messages
$MESSAGE= the message text
How is the .call file organized?
2) When an SMS is received from $SMS_CENTER2, how to get the $MESSAGE from it?
using
exten = s/${SMS_CENTER2},NoOp(${CALLERID})
exten = wait, answer
then?
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Re: [Asterisk-Users] Asterisk Crackly Bad quality
I'm interested in this, too. I find that when I use Xten or SjPhone software locally the quality is quite good, but when I use it remotely across the internet, I get quite a crackly response. *however*, if I use some SIP hardware, such as a Grandstream 236 or an IP phone (still use alaw just like Xten and SJ), the quality is great, even from halfway around the world. Literally. This leads me to think that the softphones are doing something not as well as the hardware SIP devices. Anyone have any thoughts on that? I've seen this behavior with multiple client computers, so I don't think it's just the computer that's using the softphone that's to blame... Paul - Original Message - From: Bruno Hertz [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, December 18, 2004 4:37 PM Subject: Re: [Asterisk-Users] Asterisk Crackly Bad quality On Sat, 2004-12-18 at 14:55 -0600, Steven Critchfield wrote: I highly suggest you work on getting either the RTC or USB driver loaded to provide timing if you don't already have a PSTN card for that job. OK, this is all softphones and one AVM passive BRI card here, so no digium hardware. And frankly, I'd be rather surprised if asterisk, apart from the standard kernel rtc timer, needs a special timer just to play back the demo voice and send it over the LAN. Remember, it's the initial setup we're talking about, and only the demo playback. To make sure, I compiled and loaded the ztdummy driver (from zaptel dir for 2.6 kernel). No difference. Also, if it really was the timer, that would hardly explain why e.g. FC3 and Debian Sarge behave so (wildly) different. I admit though that strange things happen sometimes :) So no, the dummy driver didn't do it. Thanks for your hints, Bruno. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for new hardware
On Sun, 2004-12-19 at 20:10 +0100, Rodolfo Grave wrote: Hi. I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm going to install: 1-)One X100P (1 FXO module) 2-)One TDM03B (3 FXO modules) I'll have the 4 FXO channels busy almost all the time, and I would like quality to be as good as possible without going to the high-level hardware. I would like to learn of some tested configurations (I've heard of problems with VIA chipsets, PCI voltages, etc). PC will be asterisk dedicated, so I'll like it to be asterisk+DigiumCards optimized. Can you recommend something? PCI voltages shouldn't be much of an issue for all FXO devices as you won't be putting any real amount of power on the lines. PCI voltages are a problem with FXS devices and the 48vDC always and [EMAIL PROTECTED] AC for ring. I would suggest something in a serverworks board. So far we have had a PIII 850 on a serverworks chipset and SCSI drive running for a long time. Our main PSTN gateway has a 418 day uptime and asterisk has been running non-stop for nearly 20 weeks. We take nearly 500 calls a day right now on that machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SMS - how to send one
The SMS in asterisk is not SMS like you're thinking... Its not for sending
to mobile phones and not something usable in the US.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Sunday, December 19, 2004 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SMS - how to send one
I've read quite a bit in the older mailing list posts and the wiki but
I'm missing some simple point.
1) What is required to send an SMS to a mobile outside the office given:
Channel: ZAP/1
send it to $SMS_RECIPIENT (which includes the final extra digit)
via
$SMS_CENTER=the national message center server for sending messages
$MESSAGE= the message text
How is the .call file organized?
2) When an SMS is received from $SMS_CENTER2, how to get the $MESSAGE from
it?
using
exten = s/${SMS_CENTER2},NoOp(${CALLERID})
exten = wait, answer
then?
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[Asterisk-Users] TE110P - problem with zone from zaptel.conf
HI, basic question. I've got a TE110P card and I'm trying to set it up with ztcfg with polish zone. ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 206: Unable to register tone zone 'pl' I've got loadzone and defaultzone set to pl, and there is a definition of that zone in zonedata.c but it doesn't work. Any hints? tia mazek -- http://www.marcinmazurek.com/ ::: nic-hdl: MM3380-RIPE GnuPG 6687 E661 98B0 AEE6 DA8B 7F48 AEE4 776F 5688 DC89 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicemailMain can't read from phone keyboard!
Wilson Pickett wrote: This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch between the phone and Asterisk. For most phones you want to use RFC2833 for both the phone and for the entry for that phone in sip.conf. Yep, and the BT will only work right with certain codecs. I think it's iLBC that suddenly won't recognize DTMF while it works with the same setting in ULAW, for example. I keep forgetting why I don't use iLBC on the BT, set it up, and then find DTMF b0rken with dtmfmode=info As most people know inband DTMF only works with the ulaw and alaw codecs. This is a codec issue, not an Asterisk issue. I thought GS fixed the need for INFO mode DTMF. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P - problem with zone from zaptel.conf
Am Sonntag, 19. Dezember 2004 21:40 schrieb Marcin Mazurek: HI, basic question. I've got a TE110P card and I'm trying to set it up with ztcfg with polish zone. ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 206: Unable to register tone zone 'pl' I've got loadzone and defaultzone set to pl, and there is a definition of that zone in zonedata.c but it doesn't work. Any hints? tia mazek voip-info.org states that some tonezones are still missing and that they are hoping that someone adds some more country specific tone zones. Jens ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztcfg seg faulting
I am running * in a development environment, adding functionality as I go. The * box has a X100P card in it which ztcfg enabled as channel 1 with fxsks signalling (fxsks=1). Everything worked fine and I was able to make inbound and outbound calls to/from the PSTN, the only issue being that some exchanges wouldn't handle the dtmf signalling, but I put that down to a peculiarity with some AU exchanges and I was able to overcome it by using pulsedial signalling. Anyway, I thought I would next slip in a PCI USB card so that I could use the USB analogue converter thingy, so I stop the * box and drop in the card and reboot. At this stage I hadn't adjusted /etc/zaptel.conf to cater for the extra channel, but when I rebooted the box ztcfg suddenly complained that it couldn't find the existing X100P card; the error message was something like: ZT_CHANCONFIG cannot config channel 1 - no such device not exactly that but you should get the idea. At this stage I went back to square one and ripped out the USB card, but to no effect, and this made me think that the X100P card had gone belly up, but why that would happen between reboots is a mystery. I then decided to put the USB card back in, comment out the reference to the fxsks signalling for the X100P on channel 1, and put in the signalling line for the USB device on channel 2 (fxoks=2). No good, ztcfg now complains that: ZT_CHANCONFIG cannot config channel 2 - no such device Now that makes me think that there might be nothing wrong with the X100P card but that the problem is more software related. I have tried recompiling the zaptel source and re-installing but to no avail. The version of zaptel is CVS which I downloaded on 3 Dec. Now, when I run ztcfg I get a seg fault. The only change that has been made to this box recently is to run up2date to update the rpm packages (Fedora Core 2) but there was nothing updated that could have impacted in the re-compile of zaptel. BTW, running kudzu on this box to discover hardware found the X100P card (which it describes as Individual Computers - Jens Schoenfeld|Intel 537) and it find the USB thingy which it describes as many things (lsusb describes it as: Bus 001 Device 002: ID 06e6:831c Tiger Jet Network, Inc. Bus 001 Device 001: ID :). Has anyone any advise on this matter. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialplan selection
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Samudra E. Haque
Sent: Sunday, December 19, 2004 12:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialplan selection
Hello,
I would like to parse inbound Asterisk IAX2 7-digit numbers
in the form of
123-4567 and strip out the first four digits, and then dial
whatever number
digits remain. If I only have three digits (000-999) and have a mix of
channels (ZAP, SIP, IAX2) could someone please point out how
I can use a
single DIAL command to just dial the extension regardless of
the type of
channel. .. For each valid extension, I have a separate dial
command anyway,
which denotes the particular channel that extension is assigned to.
I do not want to assign groups of extensions i.e., 123-A567
or 123-B567 or
123-C567 where A=ZAP, B=SIP, C=IAX2 peers respectively.
Samudra,
If I understand you correctly, you're not just looking to strip digits, but
dial an arbitrary extension without specifying the channel type in the dial
command. correct?
You should be able to accomplish this using a variable for each extension.
For example:
[globals]
X1000=SIP/1000
X1001=ZAP/1001
X1002=IAX2/1002
X1003=SIP/1003
[outbound]
exten = _123,1,Dial(${X${EXTEN:4}},10)
If the user dials 1231002, then Dial(IAX2/1002,10) should be executed.
If you have a lot of extensions then you should be able to put the variables
into a database instead.
Reid
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RE: [Asterisk-Users] dialplan selection
[globals]
X1000=SIP/1000
X1001=ZAP/1001
X1002=IAX2/1002
X1003=SIP/1003
[outbound]
exten = _123,1,Dial(${X${EXTEN:4}},10)
Oops, that line should read:
exten = _123,1,Dial(${X${EXTEN:3}},10)
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[Asterisk-Users] Can DPNSS be developed in S/w like libpri ?
Hi All, I dont know too much about the technical specs on DPNSS, but can support for it be developed in software, like libpri ? I guess what I am asking is, if DPNSS is just another signalling protocol, I suppose it can be built using software, as a layer over zaptel using a digium digital E1 card. Unless, there is something about the physical nature of the signalling, that would require different hardware ?? If its possible, has anyone thought about how to go about doing it ? Thanks Shahed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE or IAX?
On Sun, Dec 19, 2004 at 06:56:19PM +1100, Eric Bishop wrote: Hi all, Information on this topic seems a little scarce, so I thought I'd try the list Apart from the the coolness factor can anyone explain to me in what situation one would use TDMoE rather than IAX for communication betwwen 2 Asterisk servers? I thing that you're mostly better with IAX between 2 Asterisk servers. TDMoE, however, is not limited to Asterisk. It's part of zaptel. You can use it to transport a TDM link over an Ethernet network (or IP, with some kind of tunneling), and get it back as a TDM link on the other side (with proper hardware). -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP transfer(refer)
I was wondering how to make asterisk transfer a sip call automatically as sip endpoint. For example, SIP call comes to asterisk from a SIPproxy/Endpoint that offer Call Transfer feature, I want Asterisk send SIP REFER (transfer) tothat SIP proxy/Endpointso thatCaller transfersthatcall to another number. Same way as SIP IP phones offer transfer button to transfer the call. Thanks___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS - how to send one
On Sunday 19 December 2004 20:18, Brian West wrote: The SMS in asterisk is not SMS like you're thinking... Its not for sending to mobile phones and not something usable in the US. Um, sorry, but if SMS is not for sending to mobile phones, then what is it for (if it matters, I'm not in the US) ? Regards, Antony. -- Linux is going to be part of the future. It's going to be like Unix was. - Peter Moore, Asia-Pacific general manager, Microsoft Please reply to the list; please don't CC me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime and PostgreSQL
I was dloading cvs over the top of a stable branch... (Matthew told me that was a no-no...) No. That is not what I said. I said that when you do cvs update inside a previously CVS'd download of STABLE you are NOT getting the most recent version of asterisk. There are two ways to download STABLE asterisk: Grab the tarball or use CVS (cvs co -r v1-0 asterisk) If you use CVS to download the STABLE release, all future executions of cvs update will update your STABLE code only. If you go into another dir and do 'cvs co asterisk' you will get the most recent version possible. register but when I try to make a call, I get silence and some SQL failures. The SQL failures I saw where related to prepared statements. I would check to make sure you have most recent PGSql and try some test connections. I don't use PGSql so I can't help past this. -Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] TDM120 card?
This is something we would deffinatly be interested in. Our only beef with the digium cards is that you can only get 1 in a machine, unless you want to start messing with all that IRQ problems people complain about. If we want to handle 12 PRI's worth of calls, we will have to buy 3 machines ($3,500 each) and 3 4-port T1 cards ($1,500 each). Or (according to this cards manuf's) we can get 1 beefy machine and just keep adding cards until we run out of PCI slots. Has anyone actually used one of these? -Matthew - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]; Asterisk Developer Mailing List [EMAIL PROTECTED] Sent: Sunday, December 19, 2004 9:41 AM Subject: [Asterisk-Dev] TDM120 card? hi any chance of making asterisk support these? http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032 -38356249088.htm roy ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS - how to send one
between asterisk boxes and fixed line SMS I believe but never was 100% sure on this either. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Per extension/user CDR?
Me wrote: It seems that all my CDR is dumping into the Master.csv file. There is a way to create per user/extension CDR but I have looked endlessly in the Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find a way to do this.. I'm probably not the right person to answer this, but I think your idea is fantastic. It would be cool to have the ability to have a cdr file or mysql database for a given extension or a queue or even call groups. Maybe a bounty is in order, unless this is already possible but not documented :) Anyone know if this is possible with Mysql-Realtime in it's current state? Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions 1.314.480.4550 ex. 6400 1.877.999.4678 ex. 6400 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queuing
Hello, I've spent the last few days installing asterisk, and the support and documentation available here and on the wiki has been exceptional. I have now configured an E100P, with about 20 internal SIP extensions (snom 190), and a handful of international SIP extensions. Everything is working well - thank you. I now have a requirement to implement some simple call queuing. The scenario is this. This is only one operator (agent), who cannot be off-hook all the time. There are two DDIs, one is for 'valued' customers, and ensures that they are placed in the queue with a higher priority. I have achieved this already. My questions are: - At the moment, the agent must log in by dialling an extension and entering their password. Is there a way that the agent (SIP extension) can be always logged in? I guess their phone should just keep ringing if they're away from their desk. - Is there a way that the agent can be notified, maybe by a stutter tone(?), when there is a higher-priority caller in the queue? I'm still very new to asterisk, so I'm hoping that this can be achieved through the dial plan rather than a custom application. Any assistance would be greatly appreciated. Ric Searle -- Dialogue Communications Ltd. http://www.dialogue.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Per extension/user CDR?
If each account has an account code it should spawn off a CSV CDR or you can just do a mass select from SQL by account code. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM120 card?
On 2004.12.19 10:17 Eric Wieling aka ManxPower wrote: Personally I don't really approve of a company just taking Digium's design and cloning it. Huh? To what hardware are you referring? Certainly you wouldn't be indicating that the GPL only permits one licensee. http://www.zapatatelephony.org/ Or maybe you refer to some other design that is not GPLed. Lee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call screening
On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly: Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? I think I would try something like inviting the voicemail, the caller, and an auto-answer (intercom) channel on your VOIP phone into a MeetMe where your voiphone is not allowed to talk, only listen. Then you would hear what is going on and if you wanted to talk to the person you could join the MeetMe on a different line and talk to the person. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgptGZXuxZk2j.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS - how to send one
On Sunday 19 December 2004 21:35, Antony Stone wrote: On Sunday 19 December 2004 20:18, Brian West wrote: The SMS in asterisk is not SMS like you're thinking... Its not for sending to mobile phones and not something usable in the US. Um, sorry, but if SMS is not for sending to mobile phones, then what is it for (if it matters, I'm not in the US) ? Apologies for replying to my own posting, but a bit more digging has left me even more puzzled - I'm not using SMS yet, but I do plan to, and links such as http://lists.digium.com/pipermail/asterisk-cvs/2004-April/001843.html http://www.voip-info.org/wiki-Asterisk+cmd+Sms and http://www.aaisp.net.uk/aa/sms.html all seem to suggest that it can do what I want (and hope) - send receive text messages to/from standard mobile phones. Am I deluded in this hope? Antony. -- These clients are often infected by viruses or other malware and need to be fixed. If not, the user at that client needs to be fixed... - Henrik Nordstrom, on Squid users' mailing list Please reply to the list; please don't CC me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Per extension/user CDR?
I'm pretty sure if you assign account codes to your SIP and/or IAX clients in their respective .conf files then cdr files will automatically be generated for each individual account code in addition to the master. No idea about how it works with real time. hth. Aaron - Original Message - From: Matt Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, December 19, 2004 10:14 PM Subject: Re: [Asterisk-Users] Per extension/user CDR? Me wrote: It seems that all my CDR is dumping into the Master.csv file. There is a way to create per user/extension CDR but I have looked endlessly in the Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find a way to do this.. I'm probably not the right person to answer this, but I think your idea is fantastic. It would be cool to have the ability to have a cdr file or mysql database for a given extension or a queue or even call groups. Maybe a bounty is in order, unless this is already possible but not documented :) Anyone know if this is possible with Mysql-Realtime in it's current state? Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions 1.314.480.4550 ex. 6400 1.877.999.4678 ex. 6400 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 191st simultaneous call fails
Hi! Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed to create update thread! Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, span 9 Dec 14 15:44:00 WARNING[1215]: Call specified, but not found? Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9 It's not tied to which channel the call comes in on. It's some resource that's exhausted after 190 calls. A limit on threads? From what I know there is an asterisk-inherent limit of 250 (255?) Zap channels that you won't be able to surpass. I know that this doesn't explain your 191st call problem, but since you asked... :-) Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call screening
According to this it exists: http://www.voip-info.org/wiki-Asterisk+cmd+Dial However I'm testing it for the last 8 hours with no success. Recompiling after reading this: http://bugs.digium.com/bug_view_page.php?bug_id=0002905 will post back On Sun, 19 Dec 2004 14:46:01 -0800, Tracy R Reed [EMAIL PROTECTED] wrote: On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly: Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? I think I would try something like inviting the voicemail, the caller, and an auto-answer (intercom) channel on your VOIP phone into a MeetMe where your voiphone is not allowed to talk, only listen. Then you would hear what is going on and if you wanted to talk to the person you could join the MeetMe on a different line and talk to the person. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMS - how to send one
Um, sorry, but if SMS is not for sending to mobile phones, then what is it for (if it matters, I'm not in the US) ? i am in germany and use app_sms to send sms messgaes to mobile phones. app_sms does not talk directly to mobile phones but to the sms message center that in turn sends the sms to the mobile. sending sms works very well to all mobile networks (i use it for notification of voicemail new messages). receiving incoming sms is a bit more tricky as you have to send a sms message from your mobile to a non mobile number and some providers will use text to speech to read the contents of the message if they detect a fixed line number as destination number. therefore receiving sms works only if the sender uses t-online or a fixed line phone via deutsche telekom. hope that helps stefan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Per extension/user CDR?
Should be an account code field in the DB table that can be used in queries to just pull 1 accounts records ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone choices....opinion request Polycom vs Cisco
On Sun, 19 Dec 2004 12:52:40 +, w fm3 wrote: Hi I am struggling with hardware choices to get started with. My options are narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. of importance is: - functionality / integration with asterisk - headset functionality and use - voice quality - build quality Is there much of a difference between Polycom and Cisco? Scanning the group it looks like there may be slightly more issues with Polycom but I don't know how they stack up on the integration with Asterisk and future flexability. Any recommendations appreciated. Thanks Walt I would seriously start looking at IP Phones based on the PA1688 chipset. Particularly those which use one of the standard loads Have a look at http://www.aredfox.com/edownloads.htm whistl IAX2 at the time of writing is not there, its wont be long at all :-) Gary . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM120 card?
On Sun, 2004-12-19 at 14:57 -0800, Lee Howard wrote: On 2004.12.19 10:17 Eric Wieling aka ManxPower wrote: Personally I don't really approve of a company just taking Digium's design and cloning it. Huh? To what hardware are you referring? Certainly you wouldn't be indicating that the GPL only permits one licensee. http://www.zapatatelephony.org/ Or maybe you refer to some other design that is not GPLed. Or maybe you have trimmed and blown one statement out of poportion. I also despise those who wish to compete with Digium with exactly the same hardware Digium sells and advertising based on the popularity of asterisk. But as part of the line you trimmed, if someone improves or significantly change the hardware, it shows that the company/person is more than just a leach. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can DPNSS be developed in S/w like libpri ?
Shahed wrote: Hi All, I dont know too much about the technical specs on DPNSS, but can support for it be developed in software, like libpri ? I guess what I am asking is, if DPNSS is just another signalling protocol, I suppose it can be built using software, as a layer over zaptel using a digium digital E1 card. Unless, there is something about the physical nature of the signalling, that would require different hardware ?? If its possible, has anyone thought about how to go about doing it ? DPNSS runs over a D channel, just like most other ISDN related signalling protocols. However, DPNSS is a UK only protocol. It might be hard to get anyone outside the UK to take any interest in it. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call screening
Right now I'm stuck at this point:
[default]
exten = 1002,Macro(stdcs,1002,SIP/1002)
[macro-stdcs]
;; arg1 exten
;; arg2 device
exten = s,1,Wait(0.2)
exten = s,2,Playback(vm-rec-name)
exten = s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = s,4,Record(${SCREEN_FILE}:gsm|2|4)
exten = s,5,Playback(pls-wait-connect-call)
exten = s,6,Dial(${ARG2},30,gM(screen^${SCREEN_FILE}))
exten = s,7,Voicemail(u${ARG1})
exten = s,8,Playback(Goodbye)
exten = s,9,Hangup
exten = s,107,Voicemail(b${ARG1})
exten = s,108,Playback(Goodbye)
exten = s,109,Hangup
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Playback(${ARG1})
;1 TO ACCEPT, 2 TO REJECT, 3 TO TRANSFER
exten = s,3,Read(ACCEPT1|custom/2) ;this file contains the phrase'you
have an incoming call from'
exten = s,4,Noop(${ACCEPT1})
exten = s,5,Gotoif($[${ACCEPT1}=1] ?50) ;connect
exten = s,6,Gotoif($[${ACCEPT1}=2] ?30) ;reject to vm
;exten = s,6,Gotoif($[${ACCEPT1}=3] ?40) ;TRANSFER
exten = s,7,Gotoif($[${ACCEPT1}=4] ?50:50) ;any thing else connect
exten = s,30,SetVar(MACRO_RESULT=CONTINUE)
exten = s,31,System(/bin/rm ${ARG1})
;not yet written
;exten = s,40, ;ask for extension then set macro to goto that and continue
exten = s,50,System(/bin/rm ${ARG1})
when I dial exten 1002 I get the follwoing in the CLI:
-- Executing Macro(SIP/1000-906f, stdcs|1002|SIP/1002) in new stack
-- Executing Wait(SIP/1000-906f, 0.2) in new stack
-- Executing Playback(SIP/1000-906f, vm-rec-name) in new stack
-- Playing 'vm-rec-name' (language 'en')
-- Executing SetVar(SIP/1000-906f,
SCREEN_FILE=/tmp/1000-1103501744) in new stack
-- Executing Record(SIP/1000-906f,
/tmp/1000-1103501744:gsm|2|4) in new stack
-- Playing 'beep' (language 'en')
-- Executing Playback(SIP/1000-906f, pls-wait-connect-call) in
new stack-- Playing 'pls-wait-connect-call' (language 'en')
-- Executing Dial(SIP/1000-906f,
SIP/1002|30|gM(screen^/tmp/1000-1103501744)) in new stack
-- Called 1002
-- SIP/1002-1507 is ringing
-- SIP/1002-1507 answered SIP/1000-906f
-- Executing Wait(SIP/1001-1507, 0.2) in new stack
-- Executing Playback(SIP/1002-1507, /tmp/1000-1103501744) in new stack
-- Playing '/tmp/1000-1103501744' (language 'en')
-- Executing Read(SIP/1002-1507, ACCEPT1|custom/2) in new stack
-- Playing 'custom/2' (language 'en')
-- User entered ''
-- Executing NoOp(SIP/1001-1507, ) in new stack
-- Executing GotoIf(SIP/1001-1507, =1 50) in new stack
-- Executing GotoIf(SIP/1001-1507, =2 30) in new stack
-- Attempting native bridge of SIP/1000-906f and SIP/1002-1507
-- Executing VoiceMail(SIP/1002-906f, u1002) in new stack
-- Playing 'voicemail/default/1002/unavail' (language 'en')
== Spawn extension (macro-stdcs, s, 7) exited non-zero on
'SIP/1000-906f' in macro 'stdcs'
== Spawn extension (default, 1002, 1) exited non-zero on 'SIP/1000-906f'
I have no clue why the Read doesn't work, for some reason it refuses
to work from within this macro but works from any where else. Need
help ASAP.
On Sun, 19 Dec 2004 18:37:40 -0500, C F [EMAIL PROTECTED] wrote:
According to this it exists:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
However I'm testing it for the last 8 hours with no success.
Recompiling after reading this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
will post back
On Sun, 19 Dec 2004 14:46:01 -0800, Tracy R Reed
[EMAIL PROTECTED] wrote:
On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly:
Is there a way to use asterisk for call screening?
Meaning, a call comes in, asterisk answers with voicemail after I don't
pickup, and the voicemail prompt + the caller's message a played via the
sound card on asterisk. If I wan't to pick up, I do so by picking up the
phone and dialing something.
Is it doable?
I think I would try something like inviting the voicemail, the caller, and
an auto-answer (intercom) channel on your VOIP phone into a MeetMe where
your voiphone is not allowed to talk, only listen. Then you would hear
what is going on and if you wanted to talk to the person you could join
the MeetMe on a different line and talk to the person.
--
Tracy Reedhttp://copilotcom.com
This message is cryptographically signed for your protection.
Info: http://copilotconsulting.com/sig
___
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Re: [Asterisk-Users] SMS - how to send one
Could you (or anyone else who got SMS working) please send some config files? -- Socrates. On Sun, 19 Dec 2004 23:39:45 +, Stefan Reuter [EMAIL PROTECTED] wrote: Um, sorry, but if SMS is not for sending to mobile phones, then what is it for (if it matters, I'm not in the US) ? i am in germany and use app_sms to send sms messgaes to mobile phones. app_sms does not talk directly to mobile phones but to the sms message center that in turn sends the sms to the mobile. sending sms works very well to all mobile networks (i use it for notification of voicemail new messages). receiving incoming sms is a bit more tricky as you have to send a sms message from your mobile to a non mobile number and some providers will use text to speech to read the contents of the message if they detect a fixed line number as destination number. therefore receiving sms works only if the sender uses t-online or a fixed line phone via deutsche telekom. hope that helps stefan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Socrates. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: It's possible to do a codecs translation during a call in Asterisk?
Hi Matt, I have a coupple of question yet, First a couple of keys, so we know we're talking about the same things. Your setup (as I understand it) is: IAXy - Asterisk A --IAX-- Asterisk B Ok, as I see my current setup is: LANInternetLAN (IAXy A) (Asterisk A) ---IAX2-- (Asterisk B) (IAXy B) ADPCM ADPCMADPCM The easiest way would be: Asterisk A should have accounts in iax.conf for the IAXy's and the IAX link to Asterisk B. I have in the iax.conf of Asterisk A: register = userA:[EMAIL PROTECTED]:4569 and in the iax.conf of the Asterisk B: register = userB:[EMAIL PROTECTED]:4569 is that what you mean by account for the IAX link? if not please tell me the correct way to do so. In the section for the link to Asterisk B, put: disallow=all allow=gsm and do the same in Asterisk B's iax.conf file for the Asterisk A entry. That way you will end up with: IAXY --adpcm IAX-- Asterisk A --GSM IAX-- Asterisk B Asterisk A will convert from adpcm to GSM and the link between will use this. That way you have highest quality on your LAN (where bandwidth is unimportant) and then a compressed codec for traversal of the WAN (where bandwidth is obviously more important). Make sense? :-) YES! that makes a lot of sense and I think we are talking about the same think! :-D Thanks!!! -- Cheers, Matt Riddell -- Ral Gmez Cabrera ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 channel compile error
Do the paths to each of the include files exist?
If not, you will need to edit the Makefile in that directory to point
to the right include directories.
- James
On 18/12/2004, at 1:14 PM, David Adade wrote:
Hi,
Can anyone help? I get the following error when trying to complie the
h323 channel under the source installation directory
asterisk/channels/h323
i have read the readme file and kept to the recomended versions;
h.323 v1.12.2 and PWLIB v1.5.2
Thanks in advance
[EMAIL PROTECTED] h323]# make
g++ -g -c -fno-rtti -o ast_h323.o -march=i686
-DPBYTE_ORDER=PLITTLE_ENDIAN -DNDE
BUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX
-D_REENTRANT -D_GNU_S
OURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
-DPTRACING -DP_US
E_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix
-I/root/pwlib/include
-I/root/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h
323.cpp
ast_h323.cpp:35:18: h323.h: No such file or directory
ast_h323.cpp:36:21: h323pdu.h: No such file or directory
ast_h323.cpp:37:22: mediafmt.h: No such file or directory
ast_h323.cpp:38:17: lid.h: No such file or directory
In file included from ast_h323.cpp:49:
ast_h323.h:61: parse error before `{'
ast_h323.h:62: virtual outside class declaration
ast_h323.h:62: non-member function `InternalIsDescendant (const char
*)' cannot have `const' method qualifier
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:62: `H323AudioCapability' undeclared (first use this
function)
ast_h323.h:62: (Each undeclared identifier is reported only once for
each function it appears in.)
ast_h323.h:62: parse error before `::'
ast_h323.h:62: warning: no return statement in function returning
non-void
ast_h323.h: At top level:
ast_h323.h:62: virtual outside class declaration
ast_h323.h:62: non-member function `GetClass (unsigned int)' cannot
have `const' method qualifier
ast_h323.h: In function `const char *GetClass (unsigned int)':
ast_h323.h:62: parse error before `::'
ast_h323.h:62: no method `H323_G7231Capability::Class'
ast_h323.h: At top level:
ast_h323.h:62: syntax error before `('
ast_h323.h:67: syntax error before `('
ast_h323.h:69: non-member function `Clone ()' cannot have `const'
method qualifier
ast_h323.h:71: syntax error before `*'
ast_h323.h:75: non-member function `GetSubType ()' cannot have `const'
method qualifier
ast_h323.h:76: non-member function `GetFormatName ()' cannot have
`const' method qualifier
ast_h323.h:78: `H245_AudioCapability' was not declared in this scope
ast_h323.h:78: `pdu' was not declared in this scope
ast_h323.h:80: parse error before `)'
ast_h323.h:80: non-member function `OnSendingPDU (...)' cannot have
`const' method qualifier
ast_h323.h:83: parse error before `'
ast_h323.h:94: parse error before `{'
ast_h323.h:95: virtual outside class declaration
ast_h323.h:95: non-member function `InternalIsDescendant (const char
*)' cannot have `const' method qualifier
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:95: redefinition of `BOOL InternalIsDescendant (const char
*)'
ast_h323.h:62: `BOOL InternalIsDescendant (const char *)' previously
defined here
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:95: parse error before `::'
ast_h323.h:95: warning: no return statement in function returning
non-void
ast_h323.h: At top level:
ast_h323.h:95: virtual outside class declaration
ast_h323.h:95: non-member function `GetClass (unsigned int)' cannot
have `const' method qualifier
ast_h323.h: In function `const char *GetClass (unsigned int)':
ast_h323.h:95: redefinition of `const char *GetClass (unsigned int =
0)'
ast_h323.h:62: `const char *GetClass (unsigned int = 0)' previously
defined here
ast_h323.h: In function `const char *GetClass (unsigned int)':
ast_h323.h:95: parse error before `::'
ast_h323.h:95: no method `AST_G729Capability::Class'
ast_h323.h: At top level:
ast_h323.h:95: syntax error before `('
ast_h323.h:109: virtual outside class declaration
ast_h323.h:109: non-member function `Clone ()' cannot have `const'
method qualifier
ast_h323.h:116: syntax error before `*'
ast_h323.h:129: virtual outside class declaration
ast_h323.h:129: non-member function `GetSubType ()' cannot have `const'
method qualifier
ast_h323.h:133: virtual outside class declaration
ast_h323.h:133: non-member function `GetFormatName ()' cannot have
`const' method qualifier
ast_h323.h:135: parse error before `}'
ast_h323.h:141: parse error before `{'
ast_h323.h:142: virtual outside class declaration
ast_h323.h:142: non-member function `InternalIsDescendant (const char
*)' cannot have `const' method qualifier
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:142: redefinition of `BOOL InternalIsDescendant (const char
*)'
ast_h323.h:95: `BOOL InternalIsDescendant (const char *)' previously
defined here
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:142: parse
[Asterisk-Users] OT- Callwave neat app
Not sure if anyone on here has heard of this before, kind of OT but still very interesting to me and Im sure several people here. Any thoughts? http://telephonyonline.com/ar/telecom_callwave_launches_voip/index.htm Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can DPNSS be developed in S/w like libpri ?
Steve Underwood wrote: It might be hard to get anyone outside the UK to take any interest in it. You are right about that. However, if there is anyone on this list who has any thoughts on how this can be done, could you please contact me OFF list to exchange ideas ? Thanks Shahed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [Asterisk-biz] Asterisk training and certification :: AstriconTraining
I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be paying to be certified. You're pushing it here. I REFUSE TO PAY!!! I know you guys mean well but you didn't take any of us into account that know this software and know it well. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-biz- [EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Sunday, December 19, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-biz] Asterisk training and certification :: AstriconTraining *** AsteriskT Open Source Linux PBX Training and Certification Huntsville, AL and Kansas City, MO, December 20, 2004: Digium, Inc., Edvina AB and Sokol Associates today released a new program for training and certification of Asterisk professionals. Asterisk is the leading Open Source PBX for Linux, with support for both PSTN connectivity and many VoIP protocols. The first class in the Astricon Training product line is the five-day bootcamp Introduction to Asterisk. This class will be held in the US and Europe six times during 2005. The organizers and teachers is the same team that set up the Astricon 2004 conference and expo in September this year, an event that gathered over 450 Asterisk users and developers in Atlanta, GA. The new Asterisk certification is named dCAP, Digium Certified Asterisk Professional. To get the certification, one has to go through a 150 question exam as well as a practical exam, where the student builds and configures a PBX. The certification will be given by the Astricon team under license from Digium. This is an important step towards greater acceptance of Asterisk in the enterprise, says Olle E. Johansson of Edvina in Sweden. With a professional training and certification, you can ensure that your staff or your consultants has the required skills to setup and manage a mission-critical PBX platform based on Asterisk. The Asterisk Open Source project is building a professional business ecosystem, says Mark Spencer, the founder of Digium and creator of Asterisk. Many companies are now selling Asterisk-based solutions. With the 1.0 stable release in September, the Digium hardware that ranges from the IAXy end-user device to carrier-class quad-T1 cards and the Digium commercial support we have a professional platform for partnering with major enterprises. The Astricon training and dCAP certification enables us to build a network of consultants that we know will and are able to assist us working on the continued success of Asterisk. The first training class will be held in Kansas City, MO, January 17-21 2005. The cost for a five-day bootcamp with certification is $3,275 USD. Details can be found on http://www.astricon.net AsteriskT is the leading open source PBX, used all over the world. Since it is Linux-based, it inherits all of the power and stability of the operating system. Linux provides open source alternatives to proprietary applications. Asterisk is the first package to fit all telecommunication needs in a broad variety of environments. DigiumT is the creator and primary developer of Asterisk, the industry's first open source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, IP, and Ethernet architectures. Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily open source development). A full range of professional services complement these product lines, including consulting, technical support, and custom software development services. The open source communications revolution is here, and Digium is leading the way. Contacts: . Olle E. Johansson, Edvina AB, Phone +46 8 594 788 10, http://www.astricon.net . Steven M. Sokol, Sokol Associates, Phone: +1.816.822.1807, IaxTel: 700.613.9004 . Digium, press contact Rick Segrest, Phone: +1 (256) 428-6000 http://www.digium.com ___ Asterisk-Biz mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-biz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining
I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be paying to be certified. You're pushing it here. I REFUSE TO PAY!!! I know you guys mean well but you didn't take any of us into account that know this software and know it well. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-biz- [EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Sunday, December 19, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-biz] Asterisk training and certification :: AstriconTraining *** AsteriskT Open Source Linux PBX Training and Certification Huntsville, AL and Kansas City, MO, December 20, 2004: Digium, Inc., Edvina AB and Sokol Associates today released a new program for training and certification of Asterisk professionals. Asterisk is the leading Open Source PBX for Linux, with support for both PSTN connectivity and many VoIP protocols. The first class in the Astricon Training product line is the five-day bootcamp Introduction to Asterisk. This class will be held in the US and Europe six times during 2005. The organizers and teachers is the same team that set up the Astricon 2004 conference and expo in September this year, an event that gathered over 450 Asterisk users and developers in Atlanta, GA. The new Asterisk certification is named dCAP, Digium Certified Asterisk Professional. To get the certification, one has to go through a 150 question exam as well as a practical exam, where the student builds and configures a PBX. The certification will be given by the Astricon team under license from Digium. This is an important step towards greater acceptance of Asterisk in the enterprise, says Olle E. Johansson of Edvina in Sweden. With a professional training and certification, you can ensure that your staff or your consultants has the required skills to setup and manage a mission-critical PBX platform based on Asterisk. The Asterisk Open Source project is building a professional business ecosystem, says Mark Spencer, the founder of Digium and creator of Asterisk. Many companies are now selling Asterisk-based solutions. With the 1.0 stable release in September, the Digium hardware that ranges from the IAXy end-user device to carrier-class quad-T1 cards and the Digium commercial support we have a professional platform for partnering with major enterprises. The Astricon training and dCAP certification enables us to build a network of consultants that we know will and are able to assist us working on the continued success of Asterisk. The first training class will be held in Kansas City, MO, January 17-21 2005. The cost for a five-day bootcamp with certification is $3,275 USD. Details can be found on http://www.astricon.net AsteriskT is the leading open source PBX, used all over the world. Since it is Linux-based, it inherits all of the power and stability of the operating system. Linux provides open source alternatives to proprietary applications. Asterisk is the first package to fit all telecommunication needs in a broad variety of environments. DigiumT is the creator and primary developer of Asterisk, the industry's first open source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, IP, and Ethernet architectures. Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily open source development). A full range of professional services complement these product lines, including consulting, technical support, and custom software development services. The open source communications revolution is here, and Digium is leading the way. Contacts: . Olle E. Johansson, Edvina AB, Phone +46 8 594 788 10, http://www.astricon.net . Steven M. Sokol, Sokol Associates, Phone: +1.816.822.1807, IaxTel: 700.613.9004 . Digium, press contact Rick Segrest, Phone: +1 (256) 428-6000 http://www.digium.com ___ Asterisk-Biz mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-biz ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] h323 channel compile error
On Mon, 20 Dec 2004, James wrote:
Do the paths to each of the include files exist?
If not, you will need to edit the Makefile in that directory to point
to the right include directories.
- James
On 18/12/2004, at 1:14 PM, David Adade wrote:
Hi,
Can anyone help? I get the following error when trying to complie the
h323 channel under the source installation directory
asterisk/channels/h323
i have read the readme file and kept to the recomended versions;
h.323 v1.12.2 and PWLIB v1.5.2
Thanks in advance
[EMAIL PROTECTED] h323]# make
g++ -g -c -fno-rtti -o ast_h323.o -march=i686
-DPBYTE_ORDER=PLITTLE_ENDIAN -DNDE
BUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC -DP_LINUX
-D_REENTRANT -D_GNU_S
OURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
-DPTRACING -DP_US
E_PRAGMA -I../../include -I/root/pwlib/include/ptlib/unix
-I/root/pwlib/include
-I/root/openh323/include -Wno-missing-prototypes
-Wno-missing-declarations ast_h
323.cpp
ast_h323.cpp:35:18: h323.h: No such file or directory
ast_h323.cpp:36:21: h323pdu.h: No such file or directory
ast_h323.cpp:37:22: mediafmt.h: No such file or directory
ast_h323.cpp:38:17: lid.h: No such file or directory
In file included from ast_h323.cpp:49:
ast_h323.h:61: parse error before `{'
ast_h323.h:62: virtual outside class declaration
ast_h323.h:62: non-member function `InternalIsDescendant (const char
*)' cannot have `const' method qualifier
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:62: `H323AudioCapability' undeclared (first use this
function)
ast_h323.h:62: (Each undeclared identifier is reported only once for
each function it appears in.)
ast_h323.h:62: parse error before `::'
ast_h323.h:62: warning: no return statement in function returning
non-void
ast_h323.h: At top level:
ast_h323.h:62: virtual outside class declaration
ast_h323.h:62: non-member function `GetClass (unsigned int)' cannot
have `const' method qualifier
ast_h323.h: In function `const char *GetClass (unsigned int)':
ast_h323.h:62: parse error before `::'
ast_h323.h:62: no method `H323_G7231Capability::Class'
ast_h323.h: At top level:
ast_h323.h:62: syntax error before `('
ast_h323.h:67: syntax error before `('
ast_h323.h:69: non-member function `Clone ()' cannot have `const'
method qualifier
ast_h323.h:71: syntax error before `*'
ast_h323.h:75: non-member function `GetSubType ()' cannot have `const'
method qualifier
ast_h323.h:76: non-member function `GetFormatName ()' cannot have
`const' method qualifier
ast_h323.h:78: `H245_AudioCapability' was not declared in this scope
ast_h323.h:78: `pdu' was not declared in this scope
ast_h323.h:80: parse error before `)'
ast_h323.h:80: non-member function `OnSendingPDU (...)' cannot have
`const' method qualifier
ast_h323.h:83: parse error before `'
ast_h323.h:94: parse error before `{'
ast_h323.h:95: virtual outside class declaration
ast_h323.h:95: non-member function `InternalIsDescendant (const char
*)' cannot have `const' method qualifier
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:95: redefinition of `BOOL InternalIsDescendant (const char
*)'
ast_h323.h:62: `BOOL InternalIsDescendant (const char *)' previously
defined here
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
ast_h323.h:95: parse error before `::'
ast_h323.h:95: warning: no return statement in function returning
non-void
ast_h323.h: At top level:
ast_h323.h:95: virtual outside class declaration
ast_h323.h:95: non-member function `GetClass (unsigned int)' cannot
have `const' method qualifier
ast_h323.h: In function `const char *GetClass (unsigned int)':
ast_h323.h:95: redefinition of `const char *GetClass (unsigned int =
0)'
ast_h323.h:62: `const char *GetClass (unsigned int = 0)' previously
defined here
ast_h323.h: In function `const char *GetClass (unsigned int)':
ast_h323.h:95: parse error before `::'
ast_h323.h:95: no method `AST_G729Capability::Class'
ast_h323.h: At top level:
ast_h323.h:95: syntax error before `('
ast_h323.h:109: virtual outside class declaration
ast_h323.h:109: non-member function `Clone ()' cannot have `const'
method qualifier
ast_h323.h:116: syntax error before `*'
ast_h323.h:129: virtual outside class declaration
ast_h323.h:129: non-member function `GetSubType ()' cannot have `const'
method qualifier
ast_h323.h:133: virtual outside class declaration
ast_h323.h:133: non-member function `GetFormatName ()' cannot have
`const' method qualifier
ast_h323.h:135: parse error before `}'
ast_h323.h:141: parse error before `{'
ast_h323.h:142: virtual outside class declaration
ast_h323.h:142: non-member function `InternalIsDescendant (const char
*)' cannot have `const' method qualifier
ast_h323.h: In function `BOOL InternalIsDescendant (const char *)':
Re: [Asterisk-Users] [Asterisk-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining
Brian West wrote: I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be paying to be certified. You're pushing it here. Well from a newbies point of view I hope you don't pull out cause I still need help and you guys that have been around and know it backwards are a great help with setup and problems. I REFUSE TO PAY!!! I know you guys mean well but you didn't take any of us into account that know this software and know it well. I would have thought that it would be a great idea if in the process of setting this idea up they would need worldwide transer and people on the dev and long timer helpers on users list would have been prime place to find those people. Might have been an idea to come up with a Testing course first for those who think they are good enough and if they are they can pass and become the support/trainers for * in the future. Just my thoughts. And as I said above Please don't leave you guys are way to mucch support for us newbies! David bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-biz- [EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Sunday, December 19, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-biz] Asterisk training and certification :: AstriconTraining *** AsteriskT Open Source Linux PBX Training and Certification Huntsville, AL and Kansas City, MO, December 20, 2004: Digium, Inc., Edvina AB and Sokol Associates today released a new program for training and certification of Asterisk professionals. Asterisk is the leading Open Source PBX for Linux, with support for both PSTN connectivity and many VoIP protocols. The first class in the Astricon Training product line is the five-day bootcamp Introduction to Asterisk. This class will be held in the US and Europe six times during 2005. The organizers and teachers is the same team that set up the Astricon 2004 conference and expo in September this year, an event that gathered over 450 Asterisk users and developers in Atlanta, GA. The new Asterisk certification is named dCAP, Digium Certified Asterisk Professional. To get the certification, one has to go through a 150 question exam as well as a practical exam, where the student builds and configures a PBX. The certification will be given by the Astricon team under license from Digium. This is an important step towards greater acceptance of Asterisk in the enterprise, says Olle E. Johansson of Edvina in Sweden. With a professional training and certification, you can ensure that your staff or your consultants has the required skills to setup and manage a mission-critical PBX platform based on Asterisk. The Asterisk Open Source project is building a professional business ecosystem, says Mark Spencer, the founder of Digium and creator of Asterisk. Many companies are now selling Asterisk-based solutions. With the 1.0 stable release in September, the Digium hardware that ranges from the IAXy end-user device to carrier-class quad-T1 cards and the Digium commercial support we have a professional platform for partnering with major enterprises. The Astricon training and dCAP certification enables us to build a network of consultants that we know will and are able to assist us working on the continued success of Asterisk. The first training class will be held in Kansas City, MO, January 17-21 2005. The cost for a five-day bootcamp with certification is $3,275 USD. Details can be found on http://www.astricon.net AsteriskT is the leading open source PBX, used all over the world. Since it is Linux-based, it inherits all of the power and stability of the operating system. Linux provides open source alternatives to proprietary applications. Asterisk is the first package to fit all telecommunication needs in a broad variety of environments. DigiumT is the creator and primary developer of Asterisk, the industry's first open source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, IP, and Ethernet architectures. Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily open source development). A full range of professional services complement these product lines, including consulting, technical support, and custom software development services. The open source communications revolution is here, and Digium is leading the way. Contacts: . Olle E. Johansson, Edvina AB, Phone +46 8 594 788 10, http://www.astricon.net . Steven M. Sokol, Sokol Associates, Phone:
[Asterisk-Users] sip phones in different private networks have one way audio
Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call screening
OK I now know what was/is worng, my SIP is wrong it doesn't give 2 way
audio, so first I'm going to fix this and then we will see.
On Sun, 19 Dec 2004 19:26:59 -0500, C F [EMAIL PROTECTED] wrote:
Right now I'm stuck at this point:
[default]
exten = 1002,Macro(stdcs,1002,SIP/1002)
[macro-stdcs]
;; arg1 exten
;; arg2 device
exten = s,1,Wait(0.2)
exten = s,2,Playback(vm-rec-name)
exten = s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = s,4,Record(${SCREEN_FILE}:gsm|2|4)
exten = s,5,Playback(pls-wait-connect-call)
exten = s,6,Dial(${ARG2},30,gM(screen^${SCREEN_FILE}))
exten = s,7,Voicemail(u${ARG1})
exten = s,8,Playback(Goodbye)
exten = s,9,Hangup
exten = s,107,Voicemail(b${ARG1})
exten = s,108,Playback(Goodbye)
exten = s,109,Hangup
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Playback(${ARG1})
;1 TO ACCEPT, 2 TO REJECT, 3 TO TRANSFER
exten = s,3,Read(ACCEPT1|custom/2) ;this file contains the phrase'you
have an incoming call from'
exten = s,4,Noop(${ACCEPT1})
exten = s,5,Gotoif($[${ACCEPT1}=1] ?50) ;connect
exten = s,6,Gotoif($[${ACCEPT1}=2] ?30) ;reject to vm
;exten = s,6,Gotoif($[${ACCEPT1}=3] ?40) ;TRANSFER
exten = s,7,Gotoif($[${ACCEPT1}=4] ?50:50) ;any thing else connect
exten = s,30,SetVar(MACRO_RESULT=CONTINUE)
exten = s,31,System(/bin/rm ${ARG1})
;not yet written
;exten = s,40, ;ask for extension then set macro to goto that and continue
exten = s,50,System(/bin/rm ${ARG1})
when I dial exten 1002 I get the follwoing in the CLI:
-- Executing Macro(SIP/1000-906f, stdcs|1002|SIP/1002) in new stack
-- Executing Wait(SIP/1000-906f, 0.2) in new stack
-- Executing Playback(SIP/1000-906f, vm-rec-name) in new stack
-- Playing 'vm-rec-name' (language 'en')
-- Executing SetVar(SIP/1000-906f,
SCREEN_FILE=/tmp/1000-1103501744) in new stack
-- Executing Record(SIP/1000-906f,
/tmp/1000-1103501744:gsm|2|4) in new stack
-- Playing 'beep' (language 'en')
-- Executing Playback(SIP/1000-906f, pls-wait-connect-call) in
new stack-- Playing 'pls-wait-connect-call' (language 'en')
-- Executing Dial(SIP/1000-906f,
SIP/1002|30|gM(screen^/tmp/1000-1103501744)) in new stack
-- Called 1002
-- SIP/1002-1507 is ringing
-- SIP/1002-1507 answered SIP/1000-906f
-- Executing Wait(SIP/1001-1507, 0.2) in new stack
-- Executing Playback(SIP/1002-1507, /tmp/1000-1103501744) in new
stack
-- Playing '/tmp/1000-1103501744' (language 'en')
-- Executing Read(SIP/1002-1507, ACCEPT1|custom/2) in new stack
-- Playing 'custom/2' (language 'en')
-- User entered ''
-- Executing NoOp(SIP/1001-1507, ) in new stack
-- Executing GotoIf(SIP/1001-1507, =1 50) in new stack
-- Executing GotoIf(SIP/1001-1507, =2 30) in new stack
-- Attempting native bridge of SIP/1000-906f and SIP/1002-1507
-- Executing VoiceMail(SIP/1002-906f, u1002) in new stack
-- Playing 'voicemail/default/1002/unavail' (language 'en')
== Spawn extension (macro-stdcs, s, 7) exited non-zero on
'SIP/1000-906f' in macro 'stdcs'
== Spawn extension (default, 1002, 1) exited non-zero on 'SIP/1000-906f'
I have no clue why the Read doesn't work, for some reason it refuses
to work from within this macro but works from any where else. Need
help ASAP.
On Sun, 19 Dec 2004 18:37:40 -0500, C F [EMAIL PROTECTED] wrote:
According to this it exists:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
However I'm testing it for the last 8 hours with no success.
Recompiling after reading this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
will post back
On Sun, 19 Dec 2004 14:46:01 -0800, Tracy R Reed
[EMAIL PROTECTED] wrote:
On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly:
Is there a way to use asterisk for call screening?
Meaning, a call comes in, asterisk answers with voicemail after I don't
pickup, and the voicemail prompt + the caller's message a played via the
sound card on asterisk. If I wan't to pick up, I do so by picking up the
phone and dialing something.
Is it doable?
I think I would try something like inviting the voicemail, the caller, and
an auto-answer (intercom) channel on your VOIP phone into a MeetMe where
your voiphone is not allowed to talk, only listen. Then you would hear
what is going on and if you wanted to talk to the person you could join
the MeetMe on a different line and talk to the person.
--
Tracy Reedhttp://copilotcom.com
This message is cryptographically signed for your protection.
Info: http://copilotconsulting.com/sig
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
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http://lists.digium.com/mailman/listinfo/asterisk-users
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Asterisk-Users
RE: [Asterisk-Users] call screening
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, December 19, 2004 8:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call screening
OK I now know what was/is worng, my SIP is wrong it doesn't give 2 way
audio, so first I'm going to fix this and then we will see.
On Sun, 19 Dec 2004 19:26:59 -0500, C F [EMAIL PROTECTED] wrote:
Right now I'm stuck at this point:
[default]
exten = 1002,Macro(stdcs,1002,SIP/1002)
[macro-stdcs]
;; arg1 exten
;; arg2 device
exten = s,1,Wait(0.2)
exten = s,2,Playback(vm-rec-name)
exten = s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = s,4,Record(${SCREEN_FILE}:gsm|2|4)
exten = s,5,Playback(pls-wait-connect-call)
exten = s,6,Dial(${ARG2},30,gM(screen^${SCREEN_FILE}))
exten = s,7,Voicemail(u${ARG1})
exten = s,8,Playback(Goodbye)
exten = s,9,Hangup
exten = s,107,Voicemail(b${ARG1})
exten = s,108,Playback(Goodbye)
exten = s,109,Hangup
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Playback(${ARG1})
;1 TO ACCEPT, 2 TO REJECT, 3 TO TRANSFER
exten = s,3,Read(ACCEPT1|custom/2) ;this file contains the phrase'you
have an incoming call from'
exten = s,4,Noop(${ACCEPT1})
exten = s,5,Gotoif($[${ACCEPT1}=1] ?50) ;connect
exten = s,6,Gotoif($[${ACCEPT1}=2] ?30) ;reject to vm
;exten = s,6,Gotoif($[${ACCEPT1}=3] ?40) ;TRANSFER
exten = s,7,Gotoif($[${ACCEPT1}=4] ?50:50) ;any thing else connect
exten = s,30,SetVar(MACRO_RESULT=CONTINUE)
exten = s,31,System(/bin/rm ${ARG1})
;not yet written
;exten = s,40, ;ask for extension then set macro to goto that and
continue
exten = s,50,System(/bin/rm ${ARG1})
when I dial exten 1002 I get the follwoing in the CLI:
-- Executing Macro(SIP/1000-906f, stdcs|1002|SIP/1002) in new stack
-- Executing Wait(SIP/1000-906f, 0.2) in new stack
-- Executing Playback(SIP/1000-906f, vm-rec-name) in new stack
-- Playing 'vm-rec-name' (language 'en')
-- Executing SetVar(SIP/1000-906f,
SCREEN_FILE=/tmp/1000-1103501744) in new stack
-- Executing Record(SIP/1000-906f,
/tmp/1000-1103501744:gsm|2|4) in new stack
-- Playing 'beep' (language 'en')
-- Executing Playback(SIP/1000-906f, pls-wait-connect-call) in
new stack-- Playing 'pls-wait-connect-call' (language 'en')
-- Executing Dial(SIP/1000-906f,
SIP/1002|30|gM(screen^/tmp/1000-1103501744)) in new stack
-- Called 1002
-- SIP/1002-1507 is ringing
-- SIP/1002-1507 answered SIP/1000-906f
-- Executing Wait(SIP/1001-1507, 0.2) in new stack
-- Executing Playback(SIP/1002-1507, /tmp/1000-1103501744) in
new stack
-- Playing '/tmp/1000-1103501744' (language 'en')
-- Executing Read(SIP/1002-1507, ACCEPT1|custom/2) in new stack
-- Playing 'custom/2' (language 'en')
-- User entered ''
-- Executing NoOp(SIP/1001-1507, ) in new stack
-- Executing GotoIf(SIP/1001-1507, =1 50) in new stack
-- Executing GotoIf(SIP/1001-1507, =2 30) in new stack
-- Attempting native bridge of SIP/1000-906f and SIP/1002-1507
-- Executing VoiceMail(SIP/1002-906f, u1002) in new stack
-- Playing 'voicemail/default/1002/unavail' (language 'en')
== Spawn extension (macro-stdcs, s, 7) exited non-zero on
'SIP/1000-906f' in macro 'stdcs'
== Spawn extension (default, 1002, 1) exited non-zero on 'SIP/1000-
906f'
I have no clue why the Read doesn't work, for some reason it refuses
to work from within this macro but works from any where else. Need
help ASAP.
On Sun, 19 Dec 2004 18:37:40 -0500, C F [EMAIL PROTECTED] wrote:
According to this it exists:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
However I'm testing it for the last 8 hours with no success.
Recompiling after reading this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
will post back
On Sun, 19 Dec 2004 14:46:01 -0800, Tracy R Reed
[EMAIL PROTECTED] wrote:
On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly:
Is there a way to use asterisk for call screening?
Meaning, a call comes in, asterisk answers with voicemail after I
don't
pickup, and the voicemail prompt + the caller's message a played
via the
sound card on asterisk. If I wan't to pick up, I do so by picking
up the
phone and dialing something.
Is it doable?
I think I would try something like inviting the voicemail, the
caller, and
an auto-answer (intercom) channel on your VOIP phone into a MeetMe
where
your voiphone is not allowed to talk, only listen. Then you would
hear
what is going on and if you wanted to talk to the person you could
join
the MeetMe on a different line and talk to the person.
--
Tracy Reedhttp://copilotcom.com
This message is
[Asterisk-Users] one way audio on sip channels
I downloaded the latest CVS today, and since then I have only one way audio on my sip channels the callee can't hear the caller. whats wrong? I did the follwoing: cvs checkout asterisk make clean make make install running FC3 linux 2.6 64bit ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialplan help - Can dial any user but not the PSTN
What is the most efficient way to allow inbound callers to dial internal users yet restrict them from outbound PSTN calls? Today I have a basic greeting that after a welcome message allows inbound callers the ability to dial any of my users. However, it seems that since I transfer the inbound caller to a context that allows them the ability to call my internal users they have the same rights as internal users and therefore can place outbound calls. To work around this I have 2 contexts [Default] where all my users live but has an include = outbound statement. I also have a second context named [nooutbound] where I have the exact same users minus the include statement. Needless to say, I transfer my inbound callers into the [nooutbound] so they can call all my users but dont have a path to the outbound context. Works great! However, there must be a more eloquent solution without the duplication. Thoughts? Chad Brown - IdentityMine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialplan help - Can dial any user but not the PSTN
Chad Brown wrote:
What is the most efficient way to allow inbound callers to dial internal
users yet restrict them from outbound PSTN calls? Today I have a basic
greeting that after a welcome message allows inbound callers the ability
to dial any of my users. However, it seems that since I transfer the
inbound caller to a context that allows them the ability to call my
internal users they have the same rights as internal users and therefore
can place outbound calls.
To work around this I have 2 contexts... [Default] where all my users
live but has an include = outbound statement. I also have a second
context named [nooutbound] where I have the exact same users minus the
include statement. Needless to say, I transfer my inbound callers into
the [nooutbound] so they can call all my users but don't have a path to
the outbound context.
Works great! However, there must be a more eloquent solution without the
duplication. Thoughts?
try this:
[extensions]
; define your extensions here
exten = 1234,1,Dial(SIP/1234) ; John Doe's extension
exten = 5678,1,Dial(SIP/5678) ; Jane Doe's extension
[incoming]
; inbound calls end up here
include = extensions
exten = s,1,Answer
exten = s,2,Playback(greeting)
:
:
[internal]
; define your internal ppl to this context
include = extensions
exten = _9.,1,Dial(Zap/1/${EXTEN:1})
flynn
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[Asterisk-Users] iax2 event status using asterisk 1.0.3 iaxfriends
dear all, does anyone have a clue why in the event messages it show that Unregistered '1000' (AUTHENTICATED) if i'm using iaxfriends ? if using iax.conf text file configuration ... the status showed Registered '1000' (AUTHENTICATED) i'm using asterisk 1.0.3 and iaxcomm-linux (pre CVS 28 Feb 2004) regards, __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip phones in different private networks have oneway audio
Hello I have one phone (phone1) in one network, the other (phone2) in public network. both can call the other side; phone1 can be heard by phone2, phone2 can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER. Is NAT still necessary to be set on both phones? Thank you! steven So where does Asterisk fall into your equation? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT- Callwave neat app
It seems that would be pretty easy to setup with Asterisk. I wonder what amounts of usage are included at that price? Not sure if anyone on here has heard of this before, kind of OT but still very interesting to me and Im sure several people here. Any thoughts? http://telephonyonline.com/ar/telecom_callwave_launches_voip/index.htm Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT- Callwave neat app
7. How Much Does It Cost? Sign up today for a RISK-FREE 30-day trial of CallWave! Keep it, and you'll pay a special, introductory rate of only $3.95 per month. Cancel any time before your trial ends and you pay nothing. Hmm seems they aren't exactly sure what to expect. TOS didn't seem to have any usage clauses but it's only an introductory rate so when it catches on they will hike the price. =/ I agree it could probably be implimented with Asterisk too =) -- William ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for new hardware
Steven Critchfield wrote: I would suggest something in a serverworks board. So far we have had a PIII 850 on a serverworks chipset and SCSI drive running for a long time. Our main PSTN gateway has a 418 day uptime and asterisk has been running non-stop for nearly 20 weeks. We take nearly 500 calls a day right now on that machine. Are these calls using TDM400P FXO modules? Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free World Dialup and Asterisk
Hi, Julio, thanks for the tip, IAX and the incoming calls confi did the trick! FWD is up and running! THANKS! and happy holidays! Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 channel compile error
Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I HAVE THIS ERROR: [EMAIL PROTECTED] asterisk-oh323-0.7.0]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/root/asterisk-oh323-0.7.0/wrapper' ./check_ver /root/pwlib pwlib ./check_ver /root/openh323 openh323 g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapper_misc.cxx -o wrapper_misc.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapconnection.cxx -o wrapconnection.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapendpoint.cxx -o wrapendpoint.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapper.cxx -o wrapper.o wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapcaps.cxx -o wrapcaps.o touch ../asterisk-driver/chan_oh323.c g++ -DP_USE_PRAGMA -ffunction-sections -fdata-sections -D_REENTRANT -DOPENSSL_NO_KRB5 -Wall -fPIC -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\ -I/root/pwlib/include/ptlib/unix - I/root/pwlib/include -I/root/openh323/include -I/root/openh323/include/openh323 -I../asterisk-driver -c wrapgkserver.cxx -o wrapgkserver.o touch ../asterisk-driver/chan_oh323.c ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o wrapgkserver.o make[1]: Leaving directory `/root/asterisk-oh323-0.7.0/wrapper' make[1]: Entering directory `/root/asterisk-oh323-0.7.0/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/root/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1421: structure has no member named `cid' chan_oh323.c:1421: structure has no member named `cid' chan_oh323.c:1423: structure has no member named `cid' chan_oh323.c:1435: structure has no member named `cid' chan_oh323.c:1437: structure has no member named `cid' chan_oh323.c:1437: structure has no member named
[Asterisk-Users] MFC/R2 errors
Hi all I have MFCR2 successfully installed but seems to get warnings a s seen below when I start asterisk. Am running on Redhat 9. Asterisk Ready.*CLI Dec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 local_unblocking_expiredDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 far_unblocking_expired and so on -- UC channel 1 far unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Local end unblocked -- UC channel 1 local unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Far end unblocked -- UC channel 2 far unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Local end unblocked -- UC channel 2 local unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Far end unblocked -- UC channel 3 far unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Local end unblocked -- UC channel 3 local unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Far end unblocked -- UC channel 4 far unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Local end unblocked -- UC channel 4 local unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Far end unblocked -- UC channel 5 far unblockedDec 20 08:40:38 WARNING[1175077440]: chan_unicall.c:2548 handle_uc_event: UC event Local end unblocked -- UC channel 5 local unblocked ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID - TE405P - Telstra Onramp 10 - Australia
I am having problems getting incoming caller id to work on a Telstra
Onramp 10.
I have changed /DEFAULT_CIDRINGS 2/
Is there something i'm missing ?
My Cisco 7960 just shows asterisk
Thanks,
Nathan
[zapata.conf]
context=incoming
usecallingpres=yes
relaxdtmf=no
rxgain=0.0
txgain=0.0
busydetect=no
pridialplan=local
usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
adsi=no
callprogress=no
switchtype = euroisdn
signalling = pri_cpe
callgroup = 1
group = 1
immediate = yes
channel = 1-10
[extensions.conf]
[incoming]
exten = s,1,NoOp
exten = s,2,Wait(1)
exten = s,3,Answer
exten = s,4,DigitTimeout(5)
;temp addition
exten = s,6,Macro(reception,${INCOMING_DIAL})
linux*CLI show channel Zap/2-1
-- General --
Name: Zap/2-1
Type: Zap
UniqueID: 1103473308.3
Caller ID: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 1
NativeFormat: 72
WriteFormat: 8
ReadFormat: 8
1st File Descriptor: 12
Frames in: 1224
Frames out: 1986
Time to Hangup: 0
Elapsed Time: 0h0m24s
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Re: [Asterisk-Users] Caller ID - TE405P - Telstra Onramp 10 - Australia
Nathan Alberti wrote: I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed /DEFAULT_CIDRINGS 2/ Is there something i'm missing ? My Cisco 7960 just shows asterisk Thanks, Nathan SNIP linux*CLI show channel Zap/2-1 -- General -- Name: Zap/2-1 Type: Zap UniqueID: 1103473308.3 Caller ID: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 1 NativeFormat: 72 WriteFormat: 8 ReadFormat: 8 1st File Descriptor: 12 Frames in: 1224 Frames out: 1986 Time to Hangup: 0 Elapsed Time: 0h0m24s I know this might be a basic answer, but have you confirmed that CID is enabled and working on the onramp? I know when I dealt with T for an OnRamp 30 18months ago it was ordered with CID enabled but did not work for weeks when it should have. When T was chalanged about the problem it was found out that it was not enabled :( They enabled it and all the problems went away. Might be worth a thought anyway. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
