[Asterisk-Users] Re: X100P Red Alarm Ireland
Thanks for the suggestion about checking the wiring of my telephone socket! I was able to get my X100P to pass through the signal and get rid of the Red Alarm in zttool, hallelujah!!! My understanding of the problem was that the X100P wants the POTS signal on pins 2 5 whereas the Irish sockets are wired up for pins 3 4. I do have another problem with the socket that was installed by the telco (Eircom). The only way I can get the X100P to accept the signal is by connecting the POTS pair directly to the RJ11 coming from the card. If I try and go through the socket no signal gets through. I checked the connections through the socket and I have the pins wired correctly so I can only assume that the in built resistance of the socket is not letting enough current through to the card??? The socket is manufactured in the UK and has ISDN writen on it as well as having an RJ45 connector. I would love to know what is going on here... Thanks, Aaron __ Do you Yahoo!? SBC Yahoo! - Internet access at a great low price. http://promo.yahoo.com/sbc/ ___ 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] Number portability
Voicepulse connect doesn't yet offer LNP (local number portability). They said in an email that they will have LNP in 1-3 months. Does anyone know of any voip companies that DO have LNP (for US area code 504)? Thanks, Simon in New Orleans ___ 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 in the UK
On Mon, May 17, 2004 at 06:12:57PM +0100, Craig Waddington wrote: Voiptalk provide an excellent service and great support. I would hope so, as for many types of calls they're more expensive than BT's basic rates before any discounts! I'll stick to a FXO card and Telediscount/18866, ta. ___ 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] Outbound call using Soft Phone
Hello I setup Asterisk server withone single port X100P card and using X-lite soft phone. (Some how S100P USB FXS card is not working, I gave up on it) I can receive call from outsite using X100P card but I am unable to make outside call using same X100P card. I am running RedHat 9. Any idea. Thanks Deepak
RE: [Asterisk-Users] problems compiling h323 support
Le mar 18/05/2004 à 00:32, Jer a écrit : gives the same error... For what it's worth, I compiled asterisk-oh323 last week with pwlib v1.6.6-1 (Janus2) and openh323 v1.13.5-1 (Janus2) and everything compiled fine. You may want to upgrade to these versions. PS: I realize you're trying to use chan_h323 which is different. As I did not have to compile it (it came as a .deb for my system) I don't know if the problem you're running into is related, but who knows :-) ___ 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 in the UK
On Tuesday 18 May 2004 09:57, Peter Corlett wrote: On Mon, May 17, 2004 at 06:12:57PM +0100, Craig Waddington wrote: Voiptalk provide an excellent service and great support. I would hope so, as for many types of calls they're more expensive than BT's basic rates before any discounts! I'll stick to a FXO card and Telediscount/18866, ta. Ah, the indirect access ones.. may I point people in the direction of www.1899.com ? 0.5p/min anytime to the UK ... I have enquired if they offer a SIP service - but got no response :) It's a pity their website only seems to work with MSIE :( The pricing makes rather a mockery of most VoIP providers! :) Cheers, Gavin. ___ 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 in the UK
On Tue, May 18, 2004 at 10:10:49AM +0100, Gavin Hamill wrote: On Tuesday 18 May 2004 09:57, Peter Corlett wrote: [...] I'll stick to a FXO card and Telediscount/18866, ta. Ah, the indirect access ones.. may I point people in the direction of www.1899.com ? 0.5p/min anytime to the UK ... I have enquired if they offer a SIP service - but got no response :) It's a pity their website only seems to work with MSIE :( Hmm, most interesting. The rates seem identical to 18866 apart from that 0.5p/min. I must admit that 1p/min was starting to look expensive - I've been getting that rate since 1999 :) It appears to be yet another front to the Telediscount empire. The complete lack of customer service and the wxx.nl domain doing some backend stuff is a hint. The pricing makes rather a mockery of most VoIP providers! :) This isn't entirely difficult, unfortunately. ___ 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] DateTime bug?
I've just checked out the latest CVS from the 1.0-stable branch, but DateTime() seems somewhat buggy. It says something like: Tuesday May 18 11:46 AM 2004 instead of Tuesday May 18th 2004 at 11:46 AM (notice the wrong order of the words and the missing th/at) Did I miss something? Does DateTime() now take parameters that I wasn't aware of where you can tell * in what order it has to playback the date/time files? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.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] SIP in the UK
On Tue, May 18, 2004 at 10:10:49AM +0100, Gavin Hamill wrote: On Tuesday 18 May 2004 09:57, Peter Corlett wrote: [...] I'll stick to a FXO card and Telediscount/18866, ta. Ah, the indirect access ones.. may I point people in the direction of www.1899.com ? 0.5p/min anytime to the UK ... I have enquired if they offer a SIP service - but got no response :) It's a pity their website only seems to work with MSIE :( Hmm, most interesting. The rates seem identical to 18866 apart from that 0.5p/min. I must admit that 1p/min was starting to look expensive - I've been getting that rate since 1999 :) It appears to be yet another front to the Telediscount empire. The complete lack of customer service and the wxx.nl domain doing some backend stuff is a hint. The pricing makes rather a mockery of most VoIP providers! :) This isn't entirely difficult, unfortunately. But 1899 is charging 3p setup fee per call where telediscount is not? So depends how long your average call is. ___ 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 in the UK
On Tue, May 18, 2004 at 10:52:39AM +0100, Chris Stenton wrote: [...] But 1899 is charging 3p setup fee per call where telediscount is not? So depends how long your average call is. There's a 1p connection fee on 18866. So 1899 is cheaper for geographic calls longer than four minutes, and 18866 is cheaper on everything else. ___ 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] Q.931 clearing causes
Dear all - Some wisom please. We have a need to customise the Q.931 clearing causes being sent back to the network based on decisions made in some scripts or extension logic. i.e. I want to be able to decide to clear a call (either answered or during the alerting phase) with any clearing cause needed. E.g. 'All circuits busy' or 'network fault' or 'vacant number' etc. We are using Asterisk to do mobile phone testing and we need to check how our handsets behave when presented with a variety of different PSTN clearing causes. Some pointers to where in the source code we should be looking would be great. We can then make the tweaks and feed the changes back in to CVS if anyone else is interested in this feature. Best Regards Tim Robinson Motorola Ltd United Kingdom Tel. +44 1256 790472 ___ 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] Test
none of my messages are arriving on the list... just testing. ___ 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] Configure asterisk for outgoing.. need authuser parameter?
Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as: [voiptalk] type=peer secret=x username=xxx host=voiptalk.org [pipecall] type=peer secret=x username=x host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to 'Tony Hoyle sip:[EMAIL PROTECTED];tag=as4afae981' I think this means it's using the wrong username somewhere... I can dial in just fine, so it's connected.. just only one way. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ 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] Configure asterisk for outgoing.. need authuser parameter?
Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as: [voiptalk] type=peer secret=x username=xxx host=voiptalk.org [pipecall] type=peer secret=x username=x host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to 'Tony Hoyle sip:[EMAIL PROTECTED];tag=as4afae981' I think this means it's using the wrong username somewhere... I can dial in just fine, so it's connected.. just only one way. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ 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 v1.0.4.68 firmware
Thomas Gallaway -- wrote on den 17 maj 2004 16:57: Hmmm well I need to kinda figure out how to get the custom ringtones to ring on the phone... :-) ___ Asterisk-Users or how to change them /M ___ 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] No luck using asterisk as proxy...
Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I guess I didn't give enough detail in my last message, so here's as much as I've done so far: 1. I've reconfigured to network to non-NAT (was 1:1 NAT) so there's no rewriting going on. 2. I've tried various combinations of 'fromuser','fromdomain', 'username' and got nowhere. There's no authuser option on the outgoing call so this may be the issue (in which case I'll have to use a different provider as authuser!=username. Pity as they're the cheapest by far...). 3. Tried recompiling asterisk from source, just in case the debian package was broken. I still get the error: May 17 23:20:27 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to 'Tony Hoyle sip:[EMAIL PROTECTED];tag=as5c348356' Relevant chunks here of data are: [pipecall] type=peer secret= username= host=sipproxy.pipecall.com [6001] type=friend username=6001 secret= host=dynamic context=inbound-from-local The log looks like: Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A From: Tony Hoyle sip:[EMAIL PROTECTED];tag=3751201687 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1567 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103a Content-Length: 295 v=0 o=6001 8049593 8049593 IN IP4 213.208.99.115 s=X-Lite c=IN IP4 213.208.99.115 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 13 lines Using latest request as basis request Sending to 213.208.99.115 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 524302, them - 1550/0, combined - 14 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A From: Tony Hoyle sip:[EMAIL PROTECTED];tag=3751201687 To: sip:[EMAIL PROTECTED];tag=as7d10bfb2 Call-ID: [EMAIL PROTECTED] CSeq: 1567 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=7b551e23 Content-Length: 0 to 213.208.99.115:5060 sisko*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A From: Tony Hoyle sip:[EMAIL PROTECTED];tag=3751201687 To: sip:[EMAIL PROTECTED];tag=as7d10bfb2 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1567 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines sisko*CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC From: Tony Hoyle sip:[EMAIL PROTECTED];tag=3751201687 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1568 INVITE Proxy-Authorization: Digest username=6001,realm=asterisk,nonce=7b551e23,response=3f2a64418952e18bbb69bb8a5189384f,uri=sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103a Content-Length: 295 v=0 o=6001 8049593 8049593 IN IP4 213.208.99.115 s=X-Lite c=IN IP4 213.208.99.115 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Using latest request as basis request Sending to 213.208.99.115 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 524302, them - 1550/0, combined - 14 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 8378 in inbound-from-local list_route: hop: sip:[EMAIL PROTECTED]:5060 Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP
Re: [Asterisk-Users] Test
Youre making it now.. At 07:02 5/18/2004, you wrote: none of my messages are arriving on the list... just testing. ___ 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] Problems w. chan_capi + ztdummy - SNOM Problem?
A bit more clarification. If I disable alaw in asterisk but everything else as described, gsm codec is being used again. So seems like the Snom 200 got a preference for alaw even if gsm is the default and has highest priority in sip.conf. Next question is why it sounded so awfull with incoming Capi - SIP with alaw codec to the Snom. But I can live with alaw being disabled so :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lars Boegild Thomsen Sent: 18 May 2004 13:12 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems w. chan_capi + ztdummy - SNOM Problem? Actually I've played around with the last issue quite a lot and this is indeed getting weirder. Let me try to describe the problem. sip.conf is configured with: disallow = all allow = gsm allow = ulaw allow = alaw Snom phone is configured to use GSM as default codec but with Offer Answer/Full option set. If I place a call FROM the Snom phone to an external number (going out of the CAPI/Fritz/ISDN interface) everything works beautifully - and sip show channels show that the Snom phone is using GSM. If a call come IN on the Capi interface and is routed to the phone there is the described pulsating sound heard on the Snom phone alone and sip show channels report that ALAW is being used as codec. How come the choice of codec is different? AFAIK when gsm is first in sip.conf this should be the preferred codec. I haven't tried to roll back to an earlier Snom image (using 2.05d) but this problem is definitely a new one. Using an Asterisk CVS-HEAD as of today. So - I am not sure exactly where this bug is. As far as I can see there might be two problems - one that the codec of my choice is not the one being used. Second the pulsating noice when using ALAW (which should work fine too). Any ideas? Regards, Lars -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lars Boegild Thomsen Sent: 18 May 2004 12:00 To: [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] Problems w. chan_capi + ztdummy Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard to describe the sound but it breaks up so badly that it is impossible to understand the voice prompts and they also start playing extremely slow (demo congrats alone takes more than 30 seconds before going to the next prompt in the standard demo setup). I am nearly updating this particular box every day and within the last couple of days something else has happened. When dialing OUT on the ISDN card everything works fine. When someone dial IN through the card and connect to the internal Snom phone there is a pulsating background noice that can only be heard on the VoIP phone. From outside (the ISDN) things sound perfect - from inside you can still hear what is being said - but there is that pulsating quite high noice. Any ideas? Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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] Test
tmpm wrote: Youre making it now.. Sorry... I actually didn't expect it to work. My first resend from yesterday came through (twice) but my second one doesn't (including the output from sip debug) - it seems the list quietly drops long messages (14K in this case). I put it on http://www.nodomain.org/message.txt for now. Tony ___ 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
R: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?
Hi Tony, Try adding fromuser=x, maybe username= isn't enough... Just a guess, it already solved a few problems for me. -Manuel -Messaggio originale- Da: Tony Hoyle [mailto:[EMAIL PROTECTED] Inviato: martedì, 18. maggio 2004 13:03 A: [EMAIL PROTECTED] Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter? [...] [pipecall] type=peer secret=x username=x host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to 'Tony Hoyle sip:[EMAIL PROTECTED];tag=as4afae981' I think this means it's using the wrong username somewhere... I can dial in just fine, so it's connected.. just only one way. Tony ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.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] Dial and MeetMe on the same channel
Hello everybody, I would like to know whether it is possible to run Dial and MeetMe commands simultaneoously on the same channel. I am using a C AGI as below but it seems to me that only the first command that is called in the agi is executed. ... // Préparation de la commande pour l'appel du client fprintf(stderr,%s%s,numtocall, is the number to call\n); strcpy(cmd,EXEC Dial ); strcat(cmd,numtocall); //numtocall is a variable quote from teh database strcat(cmd, 60); // Exécution de la commande et libération du buffer fprintf(stderr,%s\n,cmd); printf(%s\n,cmd); fflush(stdout); resultcode = checkresult(); // Mise en conférence de l'operateur strcpy(cmd1,); strcpy(cmd1,EXEC MeetMe ); strcat(cmd1,confroom); //confroom is a variable quote from teh database strcat(cmd1,|q); fprintf(stderr,%s\n,cmd1); printf(%s\n,cmd1); fflush(stdout); .. Any reason on why only the first command is successfull?? Thanks in adavance. Lamine ___ 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] G7321 codec support in openh323
There is some sort of support of G7231 in openh323? I have seen a file in src but I think that there is only a interface not an implementation and this interface is using some itu files (free code from itu). Anyone using the G7231 in openh323? Cristian VASILIU Programator Software Principal AccessNET International SA Phone: +40.21.231.86.60 Fax: +40.21.231.86.61 Mobile: +40.788.40.14.22 E-mail: [EMAIL PROTECTED] Web: cvasiliu.vn.home.ro ___ 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: R: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?
Manuel Wenger wrote: Hi Tony, Try adding fromuser=x, maybe username= isn't enough... Just a guess, it already solved a few problems for me. I've tried fromuser=, username= and some fromdomain= combinations - unfortunately I'm not 100% sure what they change, and the error message stays the same. I think at least part of the problem is the following: Proxy-Authorization: Digest username=8378, realm=213.208.99.114, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=00fc9509e3569yx1EXM7fFX+lgZ6Byvq7g==, response=66ed3e637cb5597849619365543ee80c, opaque=MTAyNjBmOWE2MTY3MTk3MQ== The username is wrong (presumably one of the fromuser or username parameters should set that) - but which username? There are two... the phone number and the authuser. I suspect the realm name is suspect too... but I'll have to go to the CVS version for realm support if that's the issue. I've now acquired a grandstream phone so I can try that directly to the provider (specifically, sniff the packets to see what it's doing 'right' and try to make asterisk duplicate it). Tony ___ 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: Problems w. chan_capi + ztdummy - SNOM Problem?
I have this problem too. If i call out (not only with capi) the codec of my choice is used (ALAW for internal phone another for outgoing). For an incomming call alaw is used, even if i disable alaw in globals. So my bandwith is highly consumed and i can not do anything. nico Lars Boegild Thomsen wrote: Actually I've played around with the last issue quite a lot and this is indeed getting weirder. Let me try to describe the problem. sip.conf is configured with: disallow = all allow = gsm allow = ulaw allow = alaw Snom phone is configured to use GSM as default codec but with Offer Answer/Full option set. If I place a call FROM the Snom phone to an external number (going out of the CAPI/Fritz/ISDN interface) everything works beautifully - and sip show channels show that the Snom phone is using GSM. If a call come IN on the Capi interface and is routed to the phone there is the described pulsating sound heard on the Snom phone alone and sip show channels report that ALAW is being used as codec. How come the choice of codec is different? AFAIK when gsm is first in sip.conf this should be the preferred codec. I haven't tried to roll back to an earlier Snom image (using 2.05d) but this problem is definitely a new one. Using an Asterisk CVS-HEAD as of today. So - I am not sure exactly where this bug is. As far as I can see there might be two problems - one that the codec of my choice is not the one being used. Second the pulsating noice when using ALAW (which should work fine too). Any ideas? Regards, Lars -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lars Boegild Thomsen Sent: 18 May 2004 12:00 To: [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] Problems w. chan_capi + ztdummy Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard to describe the sound but it breaks up so badly that it is impossible to understand the voice prompts and they also start playing extremely slow (demo congrats alone takes more than 30 seconds before going to the next prompt in the standard demo setup). I am nearly updating this particular box every day and within the last couple of days something else has happened. When dialing OUT on the ISDN card everything works fine. When someone dial IN through the card and connect to the internal Snom phone there is a pulsating background noice that can only be heard on the VoIP phone. From outside (the ISDN) things sound perfect - from inside you can still hear what is being said - but there is that pulsating quite high noice. Any ideas? Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) Fax : +60 (3) 2057 2647 (MY) ___ 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 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 v1.0.4.68 firmware
Mikael Andersson wrote: Thomas Gallaway -- wrote on den 17 maj 2004 16:57: Hmmm well I need to kinda figure out how to get the custom ringtones to ring on the phone... :-) ___ Asterisk-Users or how to change them /M Yeah I can change them in the firmware, but I wonder if there is an option in asterisk to pass to have it do a certain ring if the call is internal, or external. The format the rings are at are after what I found out uLaw compressed 8bit 8000hz mono samples. But they also have a header infront of the file. I will play arround with it later. Maybe there is a way to chop off the header of the ones that come with it and put it infront of a regular file. -- Thomas ___ 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: Problems w. chan_capi + ztdummy - SNOM Problem?
On Tue, 18 May 2004, nicolas wrote: I have this problem too. If i call out (not only with capi) the codec of my choice is used (ALAW for internal phone another for outgoing). For an incomming call alaw is used, even if i disable alaw in globals. I'm guessing that since the incoming call is PCMA (you're on EuroISDN, right?) that greater preference is given to PCMA as the codec to use on the leg of the call to the SNOM. Notice I said guessing -- I really don't know if chan_capi works that way, but it sounds reasonable to me... So my bandwith is highly consumed and i can not do anything. You can do something: you can remove allow=alaw from the codec list for your SNOM. Granted this will affect your use of PCMA for your internal calls... If it really annoys you you will need to check out how * decides the codec to use when it's bridging from one technology to another and change it. If you're happy to use PCMA for your internal calls, why worry about it for chan_capi? You're not paying for this bandwidth :) If it's happening with incoming calls on other technologies, I think you need to look at how you specify the codecs for those other links (IAX or SIP provider). Cheers, Vic Cross ___ 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] Q.931 clearing causes
Cytowanie Robinson Tim-W10277 [EMAIL PROTECTED]: Dear all - Some wisom please. We have a need to customise the Q.931 clearing causes being sent back to the network based on decisions made in some scripts or extension logic. i.e. I want to be able to decide to clear a call (either answered or during the alerting phase) with any clearing cause needed. E.g. 'All circuits busy' or 'network fault' or 'vacant number' etc. We are using Asterisk to do mobile phone testing and we need to check how our handsets behave when presented with a variety of different PSTN clearing causes. Some pointers to where in the source code we should be looking would be great. We can then make the tweaks and feed the changes back in to CVS if anyone else is interested in this feature. channel.c ast_hangup() channels/chan_zap.c, zt_hangup() pri_disconnect takes as argument Disconnect cause so all you have to do is modify Hangup(), then ast_hangup and zt_hangup to take additonal argument that will be passed to pri_disconnect. -- Marcin Kuzmicki Agile Telecom Ltd. Golden Doors Plaza 23 6 Frederick Str. Port of Spain Trinidad and Tobago, West Indies phone (+1868) 625 20 13 fax (+1868) 624 79 88 ___ 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] speex
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 17 May 2004 18:24, James H. Cloos Jr. wrote: Just a suggestion to anyone using speex: Try running the 1.1.5 or svn code rather than 1.0.3. Is speex actually working? I was testing speex yesterday between two Asterisk'es via IAX2 and got seriously distorted sound. One server had speex-1.1.4, the other speex-1.1.5. Also, I've been unable to get below 20 for speex on a Xeon 3.02ghz. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAqgZF2TEAILET3McRAhRwAJ0dsOZRTUa/k6QTbspjq8hlPbIgnACfTho+ 2/a9wLWnTE1OiDt4n7dgFWA= =uJTK -END 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] Grandstream v1.0.4.68 firmware
I was trying to replace the header. But looks like header contains some kind of CRC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Thomas Gallaway Sent: Tuesday, May 18, 2004 3:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Grandstream v1.0.4.68 firmware Mikael Andersson wrote: Thomas Gallaway -- wrote on den 17 maj 2004 16:57: Hmmm well I need to kinda figure out how to get the custom ringtones to ring on the phone... :-) ___ Asterisk-Users or how to change them /M Yeah I can change them in the firmware, but I wonder if there is an option in asterisk to pass to have it do a certain ring if the call is internal, or external. The format the rings are at are after what I found out uLaw compressed 8bit 8000hz mono samples. But they also have a header infront of the file. I will play arround with it later. Maybe there is a way to chop off the header of the ones that come with it and put it infront of a regular file. -- Thomas ___ 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] problems with asterisk-oh323
Hello,
I've been trying to send traffic to a Cisco Call Manager 3.2, but with
no luck.
Here's whats happening:
* Call gets to CCM
* Call gets to the gateway
* Rings a couple times on destiny
* Call gets hungup.
On the CCM I get the following error: MediaManager - ERROR
wait_AuConnectErrorInd
On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not
available)
On asterisk: -- Executing Dial(SIP/test1-6e3a,
oh323/[EMAIL PROTECTED]|50|tr) in new stack
-- Called [EMAIL PROTECTED]
-- OH323/L31594 is ringing
-- H.323 call 'ip$localhost/31594' cleared, reason 1 (Cleared by
local user)-- Hungup 'OH323/L31594'
== No one is available to answer at this time
-- Executing Hangup(SIP/test1-6e3a, ) in new stack
== Spawn extension (test, h, 1) exited non-zero on 'SIP/test1-6e3a'
Attached is the trace of the asterisk-oh323 library
Any ideas on what the problem could be??
Thanks in advance
--
Pablo Endres [EMAIL PROTECTED]
ComVoz Comunications
0:01.160 OpenH323 Wrapper H323Created endpoint.
0:01.160 H323 Cleaner H323Started cleaner thread
0:01.161 OpenH323 Wrapper H323Started listener Listener[ip$*:1720]
0:01.161 OpenH323 Wrapper H323Added capability: G.729{hw} 1
0:01.161 OpenH323 Wrapper H323Added capability: UserInput/hookflash
2
0:01.161 OpenH323 Wrapper H323Added capability:
UserInput/basicString 3
0:01.161 OpenH323 Wrapper H323Added capability: UserInput/dtmf 4
0:01.161 OpenH323 Wrapper H323Added capability: UserInput/RFC2833 5
0:01.162 OpenH323 Wrapper H323UDP Binding to interface: 0.0.0.0:1
0:01.163 OpenH323 Wrapper RAS Authenticator
H235AnnexD_Procedure1no-pwd not active during GRQ SetCapability negotiation
0:01.163 OpenH323 Wrapper RAS Authenticator MD5no-pwd not active
during GRQ SetCapability negotiation
0:01.163 OpenH323 Wrapper RAS Authenticator CATno-pwd not active
during GRQ SetCapability negotiation
0:01.163 OpenH323 Wrapper H225Started gatekeeper discovery of
ip$10.0.0.2
0:01.164 OpenH323 Wrapper RAS Gatekeeper discovery on interface:
10.0.254.230:10001
0:01.164 OpenH323 Wrapper Trans Sending PDU: gatekeeperRequest 37495
0:01.164H323 Listener:9dea7d0 H323Awaiting TCP connections on port 1720
0:01.164GkMonitor:9dea3d0 RAS Background thread started
0:01.265 OpenH323 Wrapper H225RAS Receiving PDU: gatekeeperConfirm 37495
0:01.265 OpenH323 Wrapper RAS Gatekeeper discovery found
ip$216.22.64.2:1719
0:01.265 OpenH323 Wrapper RAS Gatekeeper discovered at:
216.22.64.2:1719 (if=10.0.254.230:10001)
0:01.268 OpenH323 Wrapper Trans Making request: registrationRequest
0:01.268 OpenH323 Wrapper Trans Sending PDU: registrationRequest 37496
0:01.268 OpenH323 Wrapper Trans Waiting on response to seqnum=37496
for 3.0 seconds
0:01.268 Transactor:9df23b8 Trans Starting listener thread on
Transport[remote=ip$216.22.64.2:1719 if=ip$10.0.254.230:10001]
0:01.393 Transactor:9df23b8 H225RAS Receiving PDU: registrationConfirm
37496
0:01.394 Transactor:9df23b8 RAS Registered 82BA38EC0009 with
gk1.comvoz.com
1:17.529 ThreadID=0x445c7bb0 H323Making call to: [EMAIL PROTECTED]
1:17.530 ThreadID=0x445c7bb0 H323Added capability: G.729{hw} 1
1:17.530 ThreadID=0x445c7bb0 H323Added capability: UserInput/hookflash
2
1:17.531 ThreadID=0x445c7bb0 H323Added capability:
UserInput/basicString 3
1:17.531 ThreadID=0x445c7bb0 H323Added capability: UserInput/dtmf 4
1:17.531 ThreadID=0x445c7bb0 H323Added capability: UserInput/RFC2833 5
1:17.531 ThreadID=0x445c7bb0 H323Found capability: G.729{hw} 1
1:17.531 ThreadID=0x445c7bb0 H323Found capability: UserInput/hookflash
2
1:17.531 ThreadID=0x445c7bb0 H323Found capability:
UserInput/basicString 3
1:17.531 ThreadID=0x445c7bb0 H323Found capability: UserInput/dtmf 4
1:17.531 ThreadID=0x445c7bb0 H323Found capability: UserInput/RFC2833 5
1:17.531 ThreadID=0x445c7bb0 RFC2833 Handler created
1:17.531 ThreadID=0x445c7bb0 H323Added capability: G.729{hw} 1
1:17.531 ThreadID=0x445c7bb0 H323Created new connection:
ip$localhost/18577
1:17.531 H225 Caller:9e2d880 H225Started call thread
1:17.535 H225 Caller:9e2d880 Trans Making request: admissionRequest
1:17.535 H225 Caller:9e2d880 Trans Sending PDU: admissionRequest 37497
1:17.535 H225 Caller:9e2d880 Trans Waiting on response to seqnum=37497
for 3.0 seconds
1:17.655 Transactor:9df23b8 H225RAS
Re: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940
Graham,
You need to configure something in extensions.conf to access voicemail.
I usually use something like this:
exten = 8500,1,VoiceMailMain(s${CALLERIDNUM})
exten = 8500,2,Congestion
Then you'll want to configure the voicemail URI on the 7940 so that it
calls extension 8500.
One nice thing about the Cisco phone is that they will keep track of WMI
separately for each configured line.
-brian
Graham Turner wrote:
can anyone give me a reference to the retrieval of voicemail from the
Asterisk PBX using a cisco 7940 phine running sip image.
i have configured a single voicemail box using the script, the corresponding
entry in voicemail.conf and configured the extension to use the voicemail
box .
i can see from the asterisk console the message being passed to the voice
mailbox, and correspondingly the sip phone indicates voicemail by the
flashing red on the handset and the envelope on its console
it would seem further configuration work is required to access the voice
mailbox
TIA
GT
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RE: [Asterisk-Users] quadBRI and UK ISDN2e
Hi, Maybe I have similar problem. I have a Junghanns.net quadBRI PCI Card in wiht Telefonica ISDN BRI line, and we have in zapata.conf usecallerid=yes and hidecallerid=no, but we have not the caller ID. Can I make some configuration to solve this? Thanks, Pedro -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Jon Fautley Enviado el: jueves, 08 de abril de 2004 9:51 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] quadBRI and UK ISDN2e Stephen Karrington wrote: Which brand of card did you get? The Junghanns.net quadBRI PCI Card. Just been back through BT order processing and told them to put Caller Display (as they call it) on the line, which they said they've done... getting fairly certain it's not a BT issue now :| Any help greatly appreciated, Thanks, Jon ___ 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] asterisk voicemail retrieval using a cisco 7940
Brian, thanks for your post reply .
2 further qu's if i may
in yr exten statement you use voicemailmain as the application.
i have got exten = 1001,2,Voicemail(u1001)
i know there has been recent developements to the voicemail application but
is this correct given a cvs download of early this month ??
2nd qu - where do i configure the 'voicemail uri' - have been through the
phone / line settings - or do i have to configure the SIPMAC or
sipdefault.cnf files ??
GT
- Original Message -
From: Brian Cuthie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 18, 2004 2:47 PM
Subject: Re: [Asterisk-Users] asterisk voicemail retrieval using a cisco
7940
Graham,
You need to configure something in extensions.conf to access voicemail.
I usually use something like this:
exten = 8500,1,VoiceMailMain(s${CALLERIDNUM})
exten = 8500,2,Congestion
Then you'll want to configure the voicemail URI on the 7940 so that it
calls extension 8500.
One nice thing about the Cisco phone is that they will keep track of WMI
separately for each configured line.
-brian
Graham Turner wrote:
can anyone give me a reference to the retrieval of voicemail from the
Asterisk PBX using a cisco 7940 phine running sip image.
i have configured a single voicemail box using the script, the
corresponding
entry in voicemail.conf and configured the extension to use the voicemail
box .
i can see from the asterisk console the message being passed to the voice
mailbox, and correspondingly the sip phone indicates voicemail by the
flashing red on the handset and the envelope on its console
it would seem further configuration work is required to access the voice
mailbox
TIA
GT
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[Asterisk-Users] VoIP Termination w/ 402 or 712 area code?
I realize this is a shot in the dark, but I'm trying to find a VoIP provider that offers 402 or 712 area code DID numbers. I'm almost completely convinced that no one offers these area codes (eastern Nebraska, western Iowa), however considering the wide audience of this mailing list I thought this would be a good place to ask. I would prefer a provider that allows for Asterisk use, but I realize beggars can't be choosers. -- Tony Kava Senior Network Administrator Pottawattamie County, Iowa ___ 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] asterisk voicemail retrieval using a cisco 7940
In the sip.conf file, you can specify mailbox=123 As part of the configuration for the Cisco phone. Then when there's a voicemail message in that mailbox, the red light on the phone will light up. In the Cisco's configuration you can also specify the voicemail URL as the extension you would dial to reach VoicemailMain (as defined in extensions.conf below). Then you just press the message button on the phone to retrieve your messages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, May 18, 2004 8:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940 exten = 123,1,VoiceMailMain Then dial 123 or what ever you wanna call it. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Graham Turner Sent: Tuesday, May 18, 2004 8:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940 can anyone give me a reference to the retrieval of voicemail from the Asterisk PBX using a cisco 7940 phine running sip image. i have configured a single voicemail box using the script, the corresponding entry in voicemail.conf and configured the extension to use the voicemail box . i can see from the asterisk console the message being passed to the voice mailbox, and correspondingly the sip phone indicates voicemail by the flashing red on the handset and the envelope on its console it would seem further configuration work is required to access the voice mailbox TIA GT ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.675 / Virus Database: 437 - Release Date: 5/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.675 / Virus Database: 437 - Release Date: 5/2/2004 ___ 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] VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu
option. If they just sit at the main menu, after 20 seconds, they are
transferred to the operator. If the user picks an extension from the
directory, they are transferred to the proper extension. If the called
number is not available, they are transferred into VoiceMailMain. They
leave a message, and hang up. The hang up doesn't seem to be detected in
VoiceMailMain, and they are sent back into the main incoming context of my
incoming dial plan (radiance), which after 20 seconds transfers them to an
operator. The operator answers and is greeted with the very LOUD and
annoying phone is off hook tone. If the operator hangs up, all is well,
and all the affected channels are cleared. Any tips to this? Busydetect is
NO in zapata.conf for other reasons (calls being inadvertently dropped by
asterisk).
My Dialplan:
pbxMobile*CLI show dialplan
[ Context 'default' created by 'pbx_config' ]
Include ='radiance'
[pbx_config]
Ignore pattern = '9'
[ Context 'radiance' created by 'pbx_config' ]
'9' =1. Background(radiancedirectory)
[pbx_config]
2. DigitTimeout(3)
[pbx_config]
3. ResponseTimeout(10)
[pbx_config]
'i' =1. Background(pbx-invalid)
[pbx_config]
2. Goto(radiance|s|4)
[pbx_config]
's' =1. Wait(3)
[pbx_config]
2. Answer()
[pbx_config]
3. NOOP(${CALLERID})
[pbx_config]
4. Wait(1)
[pbx_config]
5. Background(radiancewelcome)
[pbx_config]
't' =1. Playback(transferring)
[pbx_config]
2. Dial(SIP/jsantacapita|20|tT)
[pbx_config]
Include ='extensions'
[pbx_config]
[ Context 'extensions' created by 'pbx_config' ]
'.' =3. Hangup()
[pbx_config]
'0' =1. Dial(SIP/jsantacapita|20|Tt)
[pbx_config]
2. Voicemail(u100)
[pbx_config]
102. Voicemail(b100)
[pbx_config]
'100' = 1. Dial(SIP/jsantacapita|20|Tt)
[pbx_config]
2. Voicemail(u100)
[pbx_config]
102. Voicemail(b100)
[pbx_config]
'101' = 1. Dial(SIP/mthomas|20|Tt)
[pbx_config]
2. Voicemail(u101)
[pbx_config]
102. Voicemail(b101)
[pbx_config]
'102' = 1. Dial(SIP/dli|20|Tt)
[pbx_config]
2. Voicemail(u102)
[pbx_config]
102. Voicemail(b102)
[pbx_config]
'105' = 1. Dial(SIP/nmartin|20|Tt)
[pbx_config]
2. Voicemail(u105)
[pbx_config]
102. Voicemail(b105)
[pbx_config]
'600' = 1. VoiceMailMain()
[pbx_config]
'601' = 1. MeetMe()
[pbx_config]
'800' = 1. Dial(Zap/25)
[pbx_config]
2. Congestion()
[pbx_config]
'801' = 1. Dial(Zap/26)
[pbx_config]
2. Congestion()
[pbx_config]
'h' =1. Hangup()
[pbx_config]
'i' =1. Hangup()
[pbx_config]
't' =1. Hangup()
[pbx_config]
[ Context 'parkedcalls' created by 'res_parking' ]
'701' = 1. ParkedCall(701)
[res_parking]
'702' = 1. ParkedCall(702)
[res_parking]
'703' = 1. ParkedCall(703)
[res_parking]
'704' = 1. ParkedCall(704)
[res_parking]
'705' = 1. ParkedCall(705)
[res_parking]
'706' = 1. ParkedCall(706)
[res_parking]
'707' = 1. ParkedCall(707)
[res_parking]
'708' = 1. ParkedCall(708)
[res_parking]
'709' = 1. ParkedCall(709)
[res_parking]
'710' = 1. ParkedCall(710)
[res_parking]
'711' = 1. ParkedCall(711)
[res_parking]
'712' = 1. ParkedCall(712)
[res_parking]
'713' = 1. ParkedCall(713)
[res_parking]
'714' = 1. ParkedCall(714)
[res_parking]
'715' = 1. ParkedCall(715)
[res_parking]
'716' = 1. ParkedCall(716)
[res_parking]
'717' = 1. ParkedCall(717)
[res_parking]
'718' = 1. ParkedCall(718)
[res_parking]
'719' = 1. ParkedCall(719)
[res_parking]
'720' = 1. ParkedCall(720)
[res_parking]
Nik Martin
Lead Software Engineer
Radiance Technologies
[EMAIL PROTECTED]
W 251.445.0045 x105
C 251.455.4665
F 251.445.0046
___
Asterisk-Users mailing list
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RE: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940
A tip to avoid much Head-On-Desk confusion: The MWI light will only light up on cisco phones ( and all other MWI equipped phones) if the phone is in SIP context 'default' using the form: Mailbox=123 Otherwise, you must use: [EMAIL PROTECTED] I went around and around with this for 5 days until I realized that I had a group of phones in a context other than default. Nik -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Tuesday, May 18, 2004 8:59 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940 In the sip.conf file, you can specify mailbox=123 As part of the configuration for the Cisco phone. Then when there's a voicemail message in that mailbox, the red light on the phone will light up. In the Cisco's configuration you can also specify the voicemail URL as the extension you would dial to reach VoicemailMain (as defined in extensions.conf below). Then you just press the message button on the phone to retrieve your messages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, May 18, 2004 8:51 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940 exten = 123,1,VoiceMailMain Then dial 123 or what ever you wanna call it. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Graham Turner Sent: Tuesday, May 18, 2004 8:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk voicemail retrieval using a cisco 7940 can anyone give me a reference to the retrieval of voicemail from the Asterisk PBX using a cisco 7940 phine running sip image. i have configured a single voicemail box using the script, the corresponding entry in voicemail.conf and configured the extension to use the voicemail box . i can see from the asterisk console the message being passed to the voice mailbox, and correspondingly the sip phone indicates voicemail by the flashing red on the handset and the envelope on its console it would seem further configuration work is required to access the voice mailbox TIA GT ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.675 / Virus Database: 437 - Release Date: 5/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.675 / Virus Database: 437 - Release Date: 5/2/2004 ___ 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 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] problem with cdr_odbc
Hi, Has anyone made a successfull instalation of cdr_odbc?? I've install unixODBC-2.2.8 (made my own RPM) and then built the module. I'm trying to send the cdrs to a M$ SQL server. The sql connection works because I can do any query via isql. When I do the calls I get the following output on the asterisk console: -- Executing Hangup(SIP/test1-a5e1, ) in new stack == Spawn extension (test, h, 1) exited non-zero on 'SIP/test1-a5e1' cdr_odbc: Connected to SQL1 cdr_odbc: Error in Query -1 cdr_odbc: Query FAILED Call not logged! cdr_odbc: Connected to SQL1 cdr_odbc: Reconnecting to dsn SQL1 cdr_odbc: Trying Query again! cdr_odbc: Error in Query -2 cdr_odbc: Query FAILED Call not logged! -- Pablo Endres [EMAIL PROTECTED] ComVoz Comunications ___ 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] asterisk voicemail retrieval using a cisco 7940
i have got exten = 1001,2,Voicemail(u1001) This is for leaving voicemail. VoiceMailMain is for you to check voicemail. i know there has been recent developements to the voicemail application but is this correct given a cvs download of early this month ?? It hasn't changed how you check/user voicemail. 2nd qu - where do i configure the 'voicemail uri' - have been through the phone / line settings - or do i have to configure the SIPMAC or sipdefault.cnf files ?? I think you can do it via the phone.. I have always done it in the .cnf files. message_uri: xxx (xxx being the extension you use to check voicemail) bkw ___ 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] problem with cdr_odbc
Don't install from RPM or even an RPM you built. It's going to suffer the same issues no matter what if you keep using an RPM. Also what FreeTDS version are you using? == Spawn extension (default, 999, 4) exited non-zero on 'SIP/10-6e46' cdr_odbc: Query Successful! I'm using it with MySQL and MyODBC without issues. I wrote cdr_odbc.c using unixODBC 2.2.7 I don't think much as changed since then and current run it on 2.2.8 without any issues. You need to test using isql to see if you can even connect to the DSN. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pablo Endres Sent: Tuesday, May 18, 2004 9:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] problem with cdr_odbc Hi, Has anyone made a successfull instalation of cdr_odbc?? I've install unixODBC-2.2.8 (made my own RPM) and then built the module. I'm trying to send the cdrs to a M$ SQL server. The sql connection works because I can do any query via isql. When I do the calls I get the following output on the asterisk console: -- Executing Hangup(SIP/test1-a5e1, ) in new stack == Spawn extension (test, h, 1) exited non-zero on 'SIP/test1-a5e1' cdr_odbc: Connected to SQL1 cdr_odbc: Error in Query -1 cdr_odbc: Query FAILED Call not logged! cdr_odbc: Connected to SQL1 cdr_odbc: Reconnecting to dsn SQL1 cdr_odbc: Trying Query again! cdr_odbc: Error in Query -2 cdr_odbc: Query FAILED Call not logged! -- Pablo Endres [EMAIL PROTECTED] ComVoz Comunications ___ 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] G.729 on /dev/sda
I've just setup a new asterisk server, and I need to have G.729 working on this system. The problem is I don't have any IDE drives (and therefore no /dev/hda etc), but only /dev/sda. Is there really *no* way to license G.729 on a SCSI-only system? IMHO it's really stupid to replace an entire server because of a licensing issue. There *must* be a solution. Anyone, please? Or at least, is there anyone who knows who's the person (or the company) I should bother with this problem? Is it Digium or VoiceAge? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.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] VoiceMailMain dumps user back into my incoming context after leaving a message
You need to add a hangup after the VoiceMailMain I also think exten = o
will work in that case too ... not sure from VoiceMailMain but you could try
it.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nik Martin
Sent: Tuesday, May 18, 2004 9:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VoiceMailMain dumps user back into my incoming
context after leaving a message
I have a dial plan that includes a company phone directory as a main menu
option. If they just sit at the main menu, after 20 seconds, they are
transferred to the operator. If the user picks an extension from the
directory, they are transferred to the proper extension. If the called
number is not available, they are transferred into VoiceMailMain. They
leave a message, and hang up. The hang up doesn't seem to be detected in
VoiceMailMain, and they are sent back into the main incoming context of my
incoming dial plan (radiance), which after 20 seconds transfers them to an
operator. The operator answers and is greeted with the very LOUD and
annoying phone is off hook tone. If the operator hangs up, all is well,
and all the affected channels are cleared. Any tips to this? Busydetect
is
NO in zapata.conf for other reasons (calls being inadvertently dropped by
asterisk).
My Dialplan:
pbxMobile*CLI show dialplan
[ Context 'default' created by 'pbx_config' ]
Include ='radiance'
[pbx_config]
Ignore pattern = '9'
[ Context 'radiance' created by 'pbx_config' ]
'9' =1. Background(radiancedirectory)
[pbx_config]
2. DigitTimeout(3)
[pbx_config]
3. ResponseTimeout(10)
[pbx_config]
'i' =1. Background(pbx-invalid)
[pbx_config]
2. Goto(radiance|s|4)
[pbx_config]
's' =1. Wait(3)
[pbx_config]
2. Answer()
[pbx_config]
3. NOOP(${CALLERID})
[pbx_config]
4. Wait(1)
[pbx_config]
5. Background(radiancewelcome)
[pbx_config]
't' =1. Playback(transferring)
[pbx_config]
2. Dial(SIP/jsantacapita|20|tT)
[pbx_config]
Include ='extensions'
[pbx_config]
[ Context 'extensions' created by 'pbx_config' ]
'.' =3. Hangup()
[pbx_config]
'0' =1. Dial(SIP/jsantacapita|20|Tt)
[pbx_config]
2. Voicemail(u100)
[pbx_config]
102. Voicemail(b100)
[pbx_config]
'100' = 1. Dial(SIP/jsantacapita|20|Tt)
[pbx_config]
2. Voicemail(u100)
[pbx_config]
102. Voicemail(b100)
[pbx_config]
'101' = 1. Dial(SIP/mthomas|20|Tt)
[pbx_config]
2. Voicemail(u101)
[pbx_config]
102. Voicemail(b101)
[pbx_config]
'102' = 1. Dial(SIP/dli|20|Tt)
[pbx_config]
2. Voicemail(u102)
[pbx_config]
102. Voicemail(b102)
[pbx_config]
'105' = 1. Dial(SIP/nmartin|20|Tt)
[pbx_config]
2. Voicemail(u105)
[pbx_config]
102. Voicemail(b105)
[pbx_config]
'600' = 1. VoiceMailMain()
[pbx_config]
'601' = 1. MeetMe()
[pbx_config]
'800' = 1. Dial(Zap/25)
[pbx_config]
2. Congestion()
[pbx_config]
'801' = 1. Dial(Zap/26)
[pbx_config]
2. Congestion()
[pbx_config]
'h' =1. Hangup()
[pbx_config]
'i' =1. Hangup()
[pbx_config]
't' =1. Hangup()
[pbx_config]
[ Context 'parkedcalls' created by 'res_parking' ]
'701' = 1. ParkedCall(701)
[res_parking]
'702' = 1. ParkedCall(702)
[res_parking]
'703' = 1. ParkedCall(703)
[res_parking]
'704' = 1. ParkedCall(704)
[res_parking]
'705' = 1. ParkedCall(705)
[res_parking]
'706' = 1. ParkedCall(706)
[res_parking]
'707' = 1. ParkedCall(707)
[res_parking]
'708' = 1. ParkedCall(708)
[res_parking]
'709' = 1. ParkedCall(709)
[res_parking]
'710' = 1. ParkedCall(710)
[res_parking]
'711' = 1. ParkedCall(711)
[res_parking]
'712' = 1. ParkedCall(712)
[res_parking]
'713' = 1. ParkedCall(713)
[res_parking]
'714' = 1. ParkedCall(714)
[res_parking]
'715' = 1. ParkedCall(715)
[res_parking]
'716' = 1. ParkedCall(716)
[res_parking]
'717' = 1. ParkedCall(717)
[res_parking]
'718' = 1. ParkedCall(718)
[res_parking]
'719' = 1. ParkedCall(719)
[res_parking]
'720' = 1. ParkedCall(720)
[res_parking]
Nik Martin
Lead Software Engineer
Radiance Technologies
[EMAIL PROTECTED]
W 251.445.0045 x105
C 251.455.4665
F 251.445.0046
RE: [Asterisk-Users] problem with cdr_odbc
I'm using freeTDS-0.61.2, also built from RPM. Whats the deal with the RPMs? Normally I build RPMS for ALL my software, it's a lot easyer to upgrade and maintain. Isn't it a simple package of the build code? I'll try installing manually to see how it works out. On Tue, 2004-05-18 at 10:42, brian wrote: Don't install from RPM or even an RPM you built. It's going to suffer the same issues no matter what if you keep using an RPM. Also what FreeTDS version are you using? == Spawn extension (default, 999, 4) exited non-zero on 'SIP/10-6e46' cdr_odbc: Query Successful! I'm using it with MySQL and MyODBC without issues. I wrote cdr_odbc.c using unixODBC 2.2.7 I don't think much as changed since then and current run it on 2.2.8 without any issues. You need to test using isql to see if you can even connect to the DSN. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pablo Endres Sent: Tuesday, May 18, 2004 9:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] problem with cdr_odbc Hi, Has anyone made a successfull instalation of cdr_odbc?? I've install unixODBC-2.2.8 (made my own RPM) and then built the module. I'm trying to send the cdrs to a M$ SQL server. The sql connection works because I can do any query via isql. When I do the calls I get the following output on the asterisk console: -- Executing Hangup(SIP/test1-a5e1, ) in new stack == Spawn extension (test, h, 1) exited non-zero on 'SIP/test1-a5e1' cdr_odbc: Connected to SQL1 cdr_odbc: Error in Query -1 cdr_odbc: Query FAILED Call not logged! cdr_odbc: Connected to SQL1 cdr_odbc: Reconnecting to dsn SQL1 cdr_odbc: Trying Query again! cdr_odbc: Error in Query -2 cdr_odbc: Query FAILED Call not logged! -- Pablo Endres [EMAIL PROTECTED] ComVoz Comunications ___ 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 -- Pablo Endres [EMAIL PROTECTED] ComVoz Comunications ___ 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] Q.931 clearing causes
Hi, On Tue, 2004-05-18 at 06:49, Robinson Tim-W10277 wrote: Some pointers to where in the source code we should be looking would be great. We can then make the tweaks and feed the changes back in to CVS if anyone else is interested in this feature. I believe that even if there isn't a lot of interest shown for your enhancements at this point in time, Asterisk and the community at large will benefit from them in the future. So, push them up to CVS anyway! Just my opinion... :) Take care, Kanwar Systems Aligned Inc. www.systemsaligned.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] asterisk voicemail retrieval using a cisco 7940
brian wrote: i have got exten = 1001,2,Voicemail(u1001) This is for leaving voicemail. VoiceMailMain is for you to check voicemail. i know there has been recent developements to the voicemail application but is this correct given a cvs download of early this month ?? It hasn't changed how you check/user voicemail. 2nd qu - where do i configure the 'voicemail uri' - have been through the phone / line settings - or do i have to configure the SIPMAC or sipdefault.cnf files ?? I think you can do it via the phone.. I have always done it in the .cnf files. It's in the SIP Configuration part of the phone setup. (Of course this assumes you're using the SIP image.) -brian ___ 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] X100P answer in first Ring
How I can do to X100P (FXO port)answer in the first Ring? Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.
RE: [Asterisk-Users] VoiceMailMain dumps user back into my incoming context after leaving a message
Do you mean after the Voicemail (vs. after VoiceMailMain?) in each
extension?
I do have:
exten = .,3,Hangup
As step three at the bottom of my extensions context. Do I have to add it
as step 3 for every extension in the dial plan?
From my extensions.conf:
[extensions]
exten = 0,1,Dial(SIP/jsantacapita,20,Tt)
exten = 0,2,Voicemail(u100)
exten = 0,102,Voicemail(b100)
exten = 105,1,Dial(SIP/nmartin,20,Tt)
exten = 105,2,Voicemail(u105)
exten = 105,102,Voicemail(b105)
exten = 101,1,Dial(SIP/mthomas,20,Tt)
exten = 101,2,Voicemail(u101)
exten = 101,102,Voicemail(b101)
exten = 102,1,Dial(SIP/dli,20,Tt)
exten = 102,2,Voicemail(u102)
exten = 102,102,Voicemail(b102)
exten = 100,1,Dial(SIP/jsantacapita,20,Tt)
exten = 100,2,Voicemail(u100)
exten = 100,102,Voicemail(b100)
exten = .,3,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brian
Sent: Tuesday, May 18, 2004 9:45 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] VoiceMailMain dumps user back
into my incoming context after leaving a message
You need to add a hangup after the VoiceMailMain I also think
exten = o will work in that case too ... not sure from
VoiceMailMain but you could try it.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nik Martin
Sent: Tuesday, May 18, 2004 9:19 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] VoiceMailMain dumps user back into my
incoming context after leaving a message
I have a dial plan that includes a company phone directory
as a main
menu option. If they just sit at the main menu, after 20 seconds,
they are transferred to the operator. If the user picks an
extension from the
directory, they are transferred to the proper extension.
If the called
number is not available, they are transferred into VoiceMailMain.
They leave a message, and hang up. The hang up doesn't seem to be
detected in VoiceMailMain, and they are sent back into the main
incoming context of my incoming dial plan (radiance), which
after 20
seconds transfers them to an operator. The operator answers and is
greeted with the very LOUD and annoying phone is off hook
tone. If
the operator hangs up, all is well, and all the affected
channels are
cleared. Any tips to this? Busydetect is NO in
zapata.conf for other
reasons (calls being inadvertently dropped by asterisk).
My Dialplan:
pbxMobile*CLI show dialplan
[ Context 'default' created by 'pbx_config' ]
Include ='radiance'
[pbx_config]
Ignore pattern = '9'
[ Context 'radiance' created by 'pbx_config' ]
'9' =1. Background(radiancedirectory)
[pbx_config]
2. DigitTimeout(3)
[pbx_config]
3. ResponseTimeout(10)
[pbx_config]
'i' =1. Background(pbx-invalid)
[pbx_config]
2. Goto(radiance|s|4)
[pbx_config]
's' =1. Wait(3)
[pbx_config]
2. Answer()
[pbx_config]
3. NOOP(${CALLERID})
[pbx_config]
4. Wait(1)
[pbx_config]
5. Background(radiancewelcome) [pbx_config]
't' =1. Playback(transferring)
[pbx_config]
2. Dial(SIP/jsantacapita|20|tT)
[pbx_config]
Include ='extensions'
[pbx_config]
[ Context 'extensions' created by 'pbx_config' ]
'.' =3. Hangup()
[pbx_config]
'0' =1. Dial(SIP/jsantacapita|20|Tt)
[pbx_config]
2. Voicemail(u100)
[pbx_config]
102. Voicemail(b100)
[pbx_config]
'100' = 1. Dial(SIP/jsantacapita|20|Tt)
[pbx_config]
2. Voicemail(u100)
[pbx_config]
102. Voicemail(b100)
[pbx_config]
'101' = 1. Dial(SIP/mthomas|20|Tt)
[pbx_config]
2. Voicemail(u101)
[pbx_config]
102. Voicemail(b101)
[pbx_config]
'102' = 1. Dial(SIP/dli|20|Tt)
[pbx_config]
2. Voicemail(u102)
[pbx_config]
102. Voicemail(b102)
[pbx_config]
'105' = 1. Dial(SIP/nmartin|20|Tt)
[pbx_config]
2. Voicemail(u105)
[pbx_config]
102. Voicemail(b105)
[pbx_config]
'600' = 1. VoiceMailMain()
[pbx_config]
'601' = 1. MeetMe()
[pbx_config]
'800' = 1. Dial(Zap/25)
[pbx_config]
2. Congestion()
[pbx_config]
'801' = 1. Dial(Zap/26)
[pbx_config]
2. Congestion()
[pbx_config]
'h' =1. Hangup()
[pbx_config]
'i' =1. Hangup()
[pbx_config]
't' =1. Hangup()
[pbx_config]
[ Context 'parkedcalls' created by 'res_parking' ]
'701' = 1.
RE: [Asterisk-Users] X100P answer in first Ring
Title: Message put: mode=immediate in your zapata.conf file -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto SatoSent: 18 May 2004 16:14To: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P answer in first Ring How I can do to X100P (FXO port)answer in the first Ring? Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.
[Asterisk-Users] registering in sipphone
for inbound calls, i can register
context = from-sipphone
register = 1747xxx:[EMAIL PROTECTED]
but how do i configure to make outbound calls to them?
exten = _1747XXX,1,GoTo(dial-sipphone,${EXTEN},1)
[dial-sipphone]
;
; SIP to sipphone.com
;
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
^^
exten = _X.,2,Playtones(congestion)
exten = _X.,102,Playtones(busy)
exten = h,1,Hangup
randy
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Re: [Asterisk-Users] X100P answer in first Ring
On Tue, 2004-05-18 at 10:13, Alberto Sato wrote: How I can do to X100P (FXO port) answer in the first Ring? usecallerid-no -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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 error
It should be fixed in CVS -head as of sometime this morning (according to JerJer) ManxPower JerJer: Do you have any idea what's causing the 50 calls problem? JerJer fixed On Mon, 2004-05-17 at 16:46, Alberto Fernandez wrote: when asterisk has more than 50 h323 calls it craps out on me. Can anyone help? May 17 10:45:35 WARNING[1769581]: chan_sip.c:1114 create_addr: No such host: 19149191120-- Executing Dial(H323/ip$66.238.200.224:32943/16164, H323/[EMAIL PROTECTED]/1957408) in new stack-- Called [EMAIL PROTECTED]-- Executing ChanIsAvail(H323/ip$66.238.200.224:32944/16165, Sip/19149191120) in new stackMay 17 10:45:35 WARNING[1818736]: chan_sip.c:1114 create_addr: No such host: 19149191120-- Executing Dial(H323/ip$66.238.200.224:32944/16165, H323/[EMAIL PROTECTED]/1957408) in new stack-- Called [EMAIL PROTECTED] 0:15.753 H225 Caller:41c14748 assert.cxx(105) PWLib Assertion fail: Invalid array element, file /root/pwlib/include/ptlib/array.h, line 1116, Error=115 Abort, Core dump, Ignore? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA devices
Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] registering in sipphone
You simply Dial to them
exten = 2,1,dial(SIP/user,20,tr)
where user is the user registered as a peer. (in your sip.conf)
On Tue, 2004-05-18 at 11:23, Randy Bush wrote:
for inbound calls, i can register
context = from-sipphone
register = 1747xxx:[EMAIL PROTECTED]
but how do i configure to make outbound calls to them?
exten = _1747XXX,1,GoTo(dial-sipphone,${EXTEN},1)
[dial-sipphone]
;
; SIP to sipphone.com
;
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
^^
exten = _X.,2,Playtones(congestion)
exten = _X.,102,Playtones(busy)
exten = h,1,Hangup
randy
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--
Pablo Endres [EMAIL PROTECTED]
ComVoz Comunications
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Re: [Asterisk-Users] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?
Have you looked at possibly using the TDM400P with 4 FXO modules? Then you would only need to have 2 cards (currently) in the system and possibly have room for expansion in the future, if needed. That's an excellent idea, and maybe the unique way out. But, what do I do with all my X100Ps that I bought from Digium? Give them back and get my money back and buy a TDM400P(4FXO) ? :-) Isamar: I was having the same problems with a similar board. Unfortunately, there are only 4 PCI IRQs for the BIOS to assign, so SOMETHING is going to end up sharing an IRQ in your configuration, and this will break. Perhaps somebody needs to indicate this in a FAQ somewhere: X100P boards do _NOT_ play well together on the same interrupt. On my box, I was having nothing but trouble with 2 X100P boards and a TDM400P with a single FXS module. I moved cards around, and found a configuration that is problem free by avoiding the last (bottom: furthest away from the AGP) slot in the system. Since you have 5 PCI cards in your box, it sounds like you can't really do that. BTW: theoretically, there is likely a way to make Linux's APIC support force them on different APIC IRQs, but I never found a practical way to do this, nor do I know enough about APIC to even know if this would fix the problem. I don't know what to tell you, other than to echo the statement that you'll probably be better served by installing a 4 FXO TDM400P card, even though that's gonna cost you another US$400. You might try asking here on the list if anybody wants to buy some X100P boards... -fedl ___ 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] How can I dial (0 + telephone number)
I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero)to pick up the line. How can I use Dial command to dial (0 + telephone number) directly? I used exten = 10,1,Answer() exten = 10,2,Dial(Zap/1/0) exten = 10,3,Hangup It works, but I need to dial 10 and after the ring tone, the telephone number How can I do? Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.
RE: [Asterisk-Users] Q.931 clearing causes
Having looked at the latest CVS it seems that there is already some support for this. However, there are some flaws in the logic as it is currently implemented. revk put in a feature request Bug 1337 that explains the problem. We will have a look at this in the next week or so and see if it is an easy change to improve things. Rgds Tim Robinson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of M3 Freak Sent: 18 May 2004 16:04 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Q.931 clearing causes Hi, On Tue, 2004-05-18 at 06:49, Robinson Tim-W10277 wrote: Some pointers to where in the source code we should be looking would be great. We can then make the tweaks and feed the changes back in to CVS if anyone else is interested in this feature. I believe that even if there isn't a lot of interest shown for your enhancements at this point in time, Asterisk and the community at large will benefit from them in the future. So, push them up to CVS anyway! Just my opinion... :) Take care, Kanwar Systems Aligned Inc. www.systemsaligned.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 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] ATA devices
Multitech make an 8 port SIP device. Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.495.0030 f. 215.495.0031 Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. 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] X100P answer in first Ring
Title: Message I would imagine that it is I will test it , and post the result back! -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BoaterSent: 18 May 2004 16:45To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring Is this the same thing as: immediate=yes -Original Message-From: Senad Jordanovic [mailto:[EMAIL PROTECTED]Sent: Tuesday, May 18, 2004 10:21 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring put: mode=immediate in your zapata.conf file -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto SatoSent: 18 May 2004 16:14To: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P answer in first Ring How I can do to X100P (FXO port)answer in the first Ring? Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.
[Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Stephen R. Besch P.S. Grandstream, if you are listening, then Early Dial is still broken! It's been many months now to fix what apparently is just a counter bug. Come on, let's get this fixed. ___ 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 dumps user back into my incoming context after leaving a message
Nik Martin [EMAIL PROTECTED] wrote: Do you mean after the Voicemail (vs. after VoiceMailMain?) in each extension? Add a call to Hangup at the point where you'd like the call to terminate. exten = 0,1,Dial(SIP/jsantacapita,20,Tt) exten = 0,2,Voicemail(u100) exten = 0,102,Voicemail(b100) Modify your extension definition to look like this: exten = 0,1,Dial(SIP/jsantacapita,20,Tt) exten = 0,2,Voicemail(u100) exten = 0,3,Hangup exten = 0,102,Voicemail(b100) exten = 0,103,Hangup By the way, I see you're using Tt as a Dial parameter. Do you really want your incoming callers to be able to transfer the call? I imagine that someone could have fun playing with that facility. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] How can I dial (0 + telephone number)
This
is what I use. I too get dial tone to my fxo from another pbx which requires a
'9' so when you look at my config, remember that I dial a 9 to get
out.
[zap_outgoing];LOCAL CALLING STARTexten =
_XX.,1,Dial(Zap/1/9${EXTEN}#,20,r)exten =
_XX.,2,Hangup;LOCAL CALLING END
;L/D
CALLING START;ALLOWS L/D WHEN A '1' IS NOT DIALEDexten =
_NXXNXX,1,Dial(Zap/1/9${EXTEN}#,20,r)exten =
_NXXNXX,2,Hangup;ALLOWS L/D WHEN A '1' IS DIALEDexten =
_1NXXNXX,1,Dial(Zap/1/9${EXTEN:1},20,r)exten =
_1NXXNXX,2,Hangup;L/D CALLING END
;SPECIAL CALLING STARTexten =
_911,1,Dial(Zap/1/9${EXTEN}#,20,r)exten = _911,2,Hangupexten =
_411,1,Dial(Zap/1/9${EXTEN}#,20,r)exten = _411,2,Hangup;SPECIAL
CALLING END
-Original Message-From: Alberto Sato
[mailto:[EMAIL PROTECTED]Sent: Tuesday, May 18, 2004 11:02
AMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] How can I dial (0 + telephone number)
I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need
to dial 0 (zero)to pick up the line.
How can I use Dial command to dial (0 + telephone number) directly?
I used
exten = 10,1,Answer()
exten = 10,2,Dial(Zap/1/0)
exten = 10,3,Hangup
It works, but I need to dial 10 and after the ring tone, the telephone
number
How can I do?
Do you Yahoo!?SBC
Yahoo! - Internet access at a great low
price.
Re: [Asterisk-Users] ATA devices
Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, mediatrix 1124 -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] ATA devices
I have seen several vendors with this type of device. Carrier access and vpacket (now Zhone) are the two that come to mind at the moment. I think I have seen one from Audiocodes as well. ___ 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] X100P answer in first Ring
Title: Message usecallerid=no in zapata.conf -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad JordanovicSent: terça-feira, 18 de maio de 2004 13:24To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring I would imagine that it is I will test it , and post the result back! -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BoaterSent: 18 May 2004 16:45To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring Is this the same thing as: immediate=yes -Original Message-From: Senad Jordanovic [mailto:[EMAIL PROTECTED]Sent: Tuesday, May 18, 2004 10:21 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ring put: mode=immediate in your zapata.conf file -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto SatoSent: 18 May 2004 16:14To: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P answer in first Ring How I can do to X100P (FXO port)answer in the first Ring? Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.
Re: [Asterisk-Users] ATA devices
On Tue, 2004-05-18 at 09:45 -0600, Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. I was looking at the same idea this week, I wondered if I could peel of the cases of 32 GS HT286s and mount them to act like a channel bank say in two groups of 16. It would be interesting if GS would supply the cards less power supply and case. What about the IAXy? -- Dave Cotton [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] ATA devices
Check out http://www.mediatrix.com/documents/datasheets/Mediatrix_1124_0302v0.pdf Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. 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] Re: Grandstream v1.0.4.68 firmware
Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? I just slapped them all onto my TFTP server and they all load fine. Then on the bottom I can choose between the 3 ringtones and tell it to ring a certain ringtone if coming from a certain caller id. I just do not like the ringtones (piano). It'd be great if there was a way to upload own ringtones but I can not seem to be able to find out how to edit the files. ___ 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] 403 Forbidden since upgrading
Hi, I upgraded my local Asterisk (the last version was quite old), and since then, whenever anyone tries to call me via SIP/IAX thru my external Asterisk, they get 403 Forbidden as soon as I pick up. I have no trouble picking up when someone calls via PSTN. Basically, my phone (Firefly softphone) will ring when they call, but will disconnect as soon as I pick up. It won't even go to voicemail, this is from the log on the localiax; -- Accepting unauthenticated call from X.X.X.X, requested format = 4, actual format = 4 -- Executing Ringing([EMAIL PROTECTED]/16385, ) in new stack -- Executing Dial([EMAIL PROTECTED]/16385, IAX2/torh/2201|20|Ttm) in new stack -- Called torh/2201 May 18 18:02:32 WARNING[671760]: chan_iax2.c:2838 iax2_send: timestamp is 0? May 18 18:02:32 WARNING[671760]: channel.c:1445 ast_prod: Prodding channel '[EMAIL PROTECTED]/16385' failed -- Call accepted by 192.168.128.4 (format ULAW) -- Format for call is ULAW -- IAX2[torh]/6 is ringing -- Nobody picked up in 2 ms -- Hungup 'IAX2[torh]/6' -- Executing Hangup([EMAIL PROTECTED]/16385, ) in new stack == Spawn extension (iax, h, 1) exited non-zero on '[EMAIL PROTECTED]/16385' -- Hungup '[EMAIL PROTECTED]/16385' Are there any obvious places I need to look? Tor ___ 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] ATA devices
$3,000. Bummer! Todd Lieberman wrote: Multitech make an 8 port SIP device. Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.495.0030 f. 215.495.0031 Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. 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 -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.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: Grandstream v1.0.4.68 firmware
Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Didn't you hear you've gota purchase their $100,000,000 provisioning tool to enable ringtones. Jeremy McNamara ___ 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] ATA devices
Audiocodes MP124 - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 18, 2004 12:45 PM Subject: [Asterisk-Users] ATA devices Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.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 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] X100P answer in first Ring
If you have caller-id, it comes between the first and second rings. That's why * doesn't pick up until the second ring.-- Joe Dennick[EMAIL PROTECTED]-Original Message-From: Boater [EMAIL PROTECTED]Sent: Tuesday, 18. May 2004 10:45 -0500To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first RingIs this the same thing as:immediate=yes -Original Message-From: Senad Jordanovic [mailto:[EMAIL PROTECTED]Sent: Tuesday, May 18, 2004 10:21 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P answer in first Ringput:mode=immediate in your zapata.conf file-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto SatoSent: 18 May 2004 16:14To: [EMAIL PROTECTED]Subject: [Asterisk-Users] X100P answer in first RingHow I can do to X100P (FXO port) answer in the first Ring? Do you Yahoo!?SBC Yahoo! - Internet access at a great low price.
Re: [Asterisk-Users] ATA devices
Client says no LAN rewiring, so individual IAXy units might be problematic. Does anyone have IAXy pricing? Dave Cotton wrote: On Tue, 2004-05-18 at 09:45 -0600, Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. I was looking at the same idea this week, I wondered if I could peel of the cases of 32 GS HT286s and mount them to act like a channel bank say in two groups of 16. It would be interesting if GS would supply the cards less power supply and case. What about the IAXy? -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.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] ATA devices
Best price is Duct Tape, and 12 Sipura's $1200 or better $200 for a 2x16 port switches $1400 or less :) - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 18, 2004 8:45 AM Subject: [Asterisk-Users] ATA devices Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. ___ 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: Grandstream v1.0.4.68 firmware
--On Tuesday, May 18, 2004 12:30 pm -0400 Stephen R. Besch [EMAIL PROTECTED] wrote: P.S. Grandstream, if you are listening, then Early Dial is still broken! It's been many months now to fix what apparently is just a counter bug. Come on, let's get this fixed. Here, here! Iain ___ 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] ATA devices
Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. Sure...it is called Asterisk along with a TA750. Jeremy McNamara ___ 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] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. Iain ___ 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: Grandstream v1.0.4.68 firmware
Jeremy McNamara wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Didn't you hear you've gota purchase their $100,000,000 provisioning tool to enable ringtones. My ringtones just work on all the grandstream's :-) ___ 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] AArgh, * and the 7960
Out of context, this isn't much information. Is your network connection OK? Is your broadband provider having troubles? Has some upstream hardware changed that you may not be aware of? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Tuesday, May 18, 2004 1:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] AArgh, * and the 7960 I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. Iain ___ 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] AArgh, * and the 7960
Strange I do 7960 = * = IAX all day long without one jitter or any bad audio. Now if both ends are NOT running the very latest(within the last month or so) CVS-head for example if you have say a 2 month old chan_iax2.c on one end then oh boy you're in for a bad time they need to update. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Tuesday, May 18, 2004 1:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] AArgh, * and the 7960 I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. Iain ___ 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] Re: Grandstream v1.0.4.68 firmware
Thomas Gallaway wrote: My ringtones just work on all the grandstream's :-) Do the URLS for the ringtones at the top show up as something other than all zeroes? I've fiddled with this until blue in the face, and the ring sounds just like the ring it had before. This is with 1.0.4.68, but it was no different with the earlier supposedly ringtone enabled version. B. ___ 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] ATA devices
Yep, you can get one of those (MVP810), refurbished for $2K. So for the 24 ports you need, that'll be $6K + a four-port hub. Or you could get 12 Sipura SPA-2000 for $100/each, a 16-Port Switch (get something with address-cache, like the LinkSys 4116) for $100, and a few power-strips. Total is $1,300, about a quarter of the Multitech device. You can keep things simple using a TFT server for the config. If you really want to go fancy, replace the 12 individual Sipura Power-Supplies with a well-sized (65W+) 5V switched supply -- use the 5V leg of an ATX Supply (put a load on the 12V output by connecting a couple of fans). - Jay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Lieberman Sent: Tuesday, May 18, 2004 11:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ATA devices Multitech make an 8 port SIP device. Todd Lieberman http://tlsolutions.net mailto:[EMAIL PROTECTED] p. 215.495.0030 f. 215.495.0031 Michael Welter wrote: Does anyone know of a 24 port ATA device that could be installed in a phone closet? Like a channel bank, but, instead of multiplexing onto a T-1 circuit, it would convert to SIP/RTP on a LAN connection. 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 ___ 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] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. The problem has been discussed multiple times over the last several weeks. To recap, there is two things needed to incure the problem: 1. cisco 7960 phone (it discards packets with uneven timestamps) 2. asterisk had an iax problem that was fixed about a month ago assoicated with uneven timestamps. The distant iax system will need to be upgraded to fairly recent code. See previous posts for more detail. ___ 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: Grandstream v1.0.4.68 firmware
Brian Capouch wrote: Thomas Gallaway wrote: My ringtones just work on all the grandstream's :-) Do the URLS for the ringtones at the top show up as something other than all zeroes? I've fiddled with this until blue in the face, and the ring sounds just like the ring it had before. This is with 1.0.4.68, but it was no different with the earlier supposedly ringtone enabled version. *Product Model: * BT100 *Software Version: * Program--1.0.4.68Bootloader--1.0.0.16 HTML--1.0.0.31VOC--1.0.0.5 *Custom Ring Tone: * ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) -- Thomas ___ 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: Grandstream v1.0.4.68 firmware
Jeremy McNamara wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Didn't you hear you've gota purchase their $100,000,000 provisioning tool to enable ringtones. Jeremy McNamara Yes, well we can at least be thankful that they haven't adopted the Cisco model. Then we would need to pay them a licensing fee each time the phone rings. SRB ___ 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 announce?
I use the Zultys 4x4 and it will allow me to announce before it transfers. But he GS BT-100 just transfers it right away. Kyle brian k. west wrote: No way to do that without writing your own custom application to do it. bkw - Original Message - From: Gavin Hollinger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 11:22 PM Subject: Re: [Asterisk-Users] call announce? exten = s,5,ParkAndAnnounce Yeah I need to elaborate on what I am trying to do. Sorry. Find Me Follow me. John calls Peter and records his name. John gets parked and listens to music. Then I call several possible locations for Peter, simultaneously or one by one as a selectable option. When you dial a location looking for Peter, I want to play a recording something like this: Hello 'John' holding for 'Peter' to connect the call press 1, to send to voice mail, press 2, to keep looking, press 3 When 1 is pressed the call is un-parked and John and Peter talk, while peter has the power to transfer John to another number, etc. When 2 is pressed the call goes to voice mail. When 3 is pressed, the next possible location for 'Peter' is tried. I ONLY want those 3 options, I don't want people connecting to other peoples parked calls etc. Thanks Gavin ___ 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 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] blocked caller id
I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says caller id unavialable. With the spelling done right Any ideas on how this wold be accomplished? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x102 ___ 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] TDM400P and AGGRESSIVE_SUPPRESSOR dropping calls
Hi, I had been using 4 X100P cards in my Asterisk box, but 2 of them were sharing an interrupt. Therefore, periodically I would hear beeps and clicks that I had assumed were a result of this. So, I ordered a TDM400P with 4 FXO modules and installed it in the box last night. Today, we've had nothing but problems with it dropping calls. I installed the latest CVS of everything, and we've been getting random hangups. If I disable AGGRESSIVE_SUPPRESSOR, the random hangups seem to stop but we of course experience *really* bad echo. I have busydetect=yes and busycount=8, which has previously been working just fine with the X100Ps. Does anyone have an idea what's going on or how to fix 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
[Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Thomas Gallaway wrote: Jeremy McNamara wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? Didn't you hear you've gota purchase their $100,000,000 provisioning tool to enable ringtones. My ringtones just work on all the grandstream's :-) Bizarre, really bizarre. SRB ___ 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: Grandstream v1.0.4.68 firmware
Thomas Galloway wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? I just slapped them all onto my TFTP server and they all load fine. Then on the bottom I can choose between the 3 ringtones and tell it to ring a certain ringtone if coming from a certain caller id. I just do not like the ringtones (piano). It'd be great if there was a way to upload own ringtones but I can not seem to be able to find out how to edit the files. That has not been my experience. I do indeed get the option to associate each ring tone with a given caller ID (which in my estimation is a really stupid implementation anyway - the real value would be in associating each line on the 2 line GS with a different ring tone. The caller ID already tells you who is calling). However, I can put anything I want into the text boxes and nothing happens - I always get the system ring tone. And, what are those stupid little radio boxes for. No matter which one I check, when the screen refreshes it defaults back to the System Ring Tone. Here's what they look like (the o's are supposed to be radio boxes): o System Ring Tone o Custom Ring tone 1, used if incoming caller ID is (Text Box) o Custom Ring tone 2, etc. What's this supposed to mean? It implies that if I select one of the custom ring tones, then the phone will ring on the matching CID, otherwise, it won't ring at all! This feature really needs work. I hope it doesn't wind up like the useless Daylight Savings Time option, which you may have noticed does not pay any attention to the date so you have to log into each and every phone and change the option anyway (please, correct me if this has been fixed). Why bother? I can just as easily change the time zone and get the same effect. GS is obviously targeting their phones for the consumer market where a customer has 1, maybe 2 phones and this kind of thing is irrelevant. The whole concept of their overpriced provisioning system is rather a funny joke in this context, and a rather pretentious one at that. SRB ___ 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] ATA devices
Yep, you can get one of those (MVP810), refurbished for $2K. So for the 24 ports you need, that'll be $6K + a four-port hub. Or you could get 12 Sipura SPA-2000 for $100/each, a 16-Port Switch (get something with address-cache, like the LinkSys 4116) for $100, and a few power-strips. Total is $1,300, about a quarter of the Multitech device. You can keep things simple using a TFT server for the config. If you really want to go fancy, replace the 12 individual Sipura Power-Supplies with a well-sized (65W+) 5V switched supply -- use the 5V leg of an ATX Supply (put a load on the 12V output by connecting a couple of fans). Is it really a problem commandeering a cat5 pair in the closet for a T1 and using a cheap AB1 (FXS, right?) and plug the other end into your * box? It seems like a colossal wiring and yucky configuration/maintenance mess otherwise. AB1 with 24 FXS under $800 on ebay T100P: $500 new D50 to BIX: $30 Total: ~$1330 or less. -A. ___ 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] MeetMe conference delay increasing
I've just noticed a strange behaviour with a MeetMe conference. I have a pair of phones (GS BT102) on my desk, and dialled both of them into a conference on speakerphone. If I spoke or made a sound, I heard it replayed from both speakers together a split second later, as expected. I went away for about 15 minutes, leaving the conference running. When I came back any sound I made came back out of the speakers, not immediately, but with about a 2 second delay or more! I hung up one of the phones and redialled the conference. Now a sound was relayed out of the redialled phone immediately, but still delayed on the other phone. Hanging up that other phone and redialling restored the immediate timing on that phone too. I was using the iLBC codec on both phones, so I disabled ilbc in sip.conf and then retried the test using alaw instead. Leaving the conference running for an hour didn't introduce any appreciable delay. So the increasing delay must be due to the iLBC codec in either the phone or in Asterisk. For a final test, I re-enabled iLBC in the phones, and changed the iLBC frame size from 20ms to 30ms. I then repeated the test. The delay still increased over time, but not as quickly. Comments? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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
[Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
Thomas Gallaway wrote: Brian Capouch wrote: Thomas Gallaway wrote: My ringtones just work on all the grandstream's :-) Do the URLS for the ringtones at the top show up as something other than all zeroes? I've fiddled with this until blue in the face, and the ring sounds just like the ring it had before. This is with 1.0.4.68, but it was no different with the earlier supposedly ringtone enabled version. *Product Model: * BT100 *Software Version: * Program--1.0.4.68Bootloader--1.0.0.16 HTML--1.0.0.31VOC--1.0.0.5 *Custom Ring Tone: * ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) -- Thomas Well in my case, the ring versions are all 0.0.0.0 no matter what I do. Could you also post the exact spelling of the binarys on the tftp, including capitalization, access rights and access mode. Mine are ring1.bin, ring2.bin, ring3.bin, all lower case, owned by root, in the asterisk group and with r/w mode=rw_rw_r__ (664), which is the same as all the other items on the server, which do in fact get loaded. Also, I've checked the binaries and they do in fact have version 1.0.0.0 embedded in the file (look at hex offset 6, which is where the version signature of all the GS bianries is located). I also can connect to the tftp server from another machine and successfully get ring1.bin. Perhaps the binaries that came with my copy of the firmware are corrupted. Maybe you could zip up the tones you are using and post them to me so I could see if these fix the problem. SRB ___ 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] speex
-fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse -msse2 -mfpmath=sse Made no difference whatsoever on my P4/Xeon 2.4GHz for iLBC (I don't use speex). -A. ___ 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] blocked caller id
check the caller id in your incoming extension before you pass to to a end user. Reset $calleridname to unavaliable if no number is given On Tue, 2004-05-18 at 15:18, Roger wrote: I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says caller id unavialable. With the spelling done right Any ideas on how this wold be accomplished? ___ 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 announce?
On Tue, 2004-05-18 at 14:11, Kyle Hagan wrote: I use the Zultys 4x4 and it will allow me to announce before it transfers. But he GS BT-100 just transfers it right away. Supervised aka consultative transfer is where you get to talk to the person you are transfering to before you complete the transfer. Blind transfer is when you transfer the call without talking to the destination. Asterisk's Call Parking requires consultative transfers or you will never hear the parking extension reads back to you. Generally speaking a consultative transfer is a special case of 3-way calling As you can see at http://www.grandstream.com/Product_Spec.pdf the BT102D is the only GS phone that supports 3-way calling. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] snom 200 phones.
I have about 5 snom 200 phones working fine with everything. Voicemail, Transfers and all. Except I can't seem to use them to pickup parked calls nor place a call on park. I also have sipura-2000 with analog phones that are able to pickup parked calls and to park them. Most of them are on firmware 2.04g I have upgraded one to 2.05c for testing but this did not fix the problem. I get no error message on the CLI and I am at a lost of where I can begin to look for a problem. I have other Sip phones working fine. Cisco 7960'g, IpDialogs They all work fine. ATA 186 and Sipura-2000 are also working fine they all can park a call and pick them up. - \ \\_ Ariel Batista // / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ 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 announce?
I had to write my own Dial2 application to do this, which is a copy of the app_dial.c source with this feature added. I didn't have it record the incoming caller's name, but rather prompt the answering user as to whether or not to accept the call. It would be trivial using extension logic to have the call answered, prompt the user to say their name, and then announce that with the call (using the A option with the dial command) the rest you would have to customize yourself. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Tuesday, May 18, 2004 3:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] call announce? I use the Zultys 4x4 and it will allow me to announce before it transfers. But he GS BT-100 just transfers it right away. Kyle brian k. west wrote: No way to do that without writing your own custom application to do it. bkw - Original Message - From: Gavin Hollinger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 17, 2004 11:22 PM Subject: Re: [Asterisk-Users] call announce? exten = s,5,ParkAndAnnounce Yeah I need to elaborate on what I am trying to do. Sorry. Find Me Follow me. John calls Peter and records his name. John gets parked and listens to music. Then I call several possible locations for Peter, simultaneously or one by one as a selectable option. When you dial a location looking for Peter, I want to play a recording something like this: Hello 'John' holding for 'Peter' to connect the call press 1, to send to voice mail, press 2, to keep looking, press 3 When 1 is pressed the call is un-parked and John and Peter talk, while peter has the power to transfer John to another number, etc. When 2 is pressed the call goes to voice mail. When 3 is pressed, the next possible location for 'Peter' is tried. I ONLY want those 3 options, I don't want people connecting to other peoples parked calls etc. Thanks Gavin ___ 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 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] Re: Grandstream v1.0.4.68 firmware
Stephen R. Besch wrote: Thomas Galloway wrote: Stephen R. Besch wrote: Duane wrote: Grandstream v1.0.4.68 firmware Am I missing something obvious about the new ringtone feature? The 4.68 firmware updates as usual from my TFTP server, the new version shows up in the phone's web page, but the ring tones, while present on the server and referenced on the web page, all show a version of 0.0.0.0, and all functionality regarding them is disabled. Are we maybe jumping the gun here a little bit or is there something special about getting them to load? I just slapped them all onto my TFTP server and they all load fine. Then on the bottom I can choose between the 3 ringtones and tell it to ring a certain ringtone if coming from a certain caller id. I just do not like the ringtones (piano). It'd be great if there was a way to upload own ringtones but I can not seem to be able to find out how to edit the files. That has not been my experience. I do indeed get the option to associate each ring tone with a given caller ID (which in my estimation is a really stupid implementation anyway - the real value would be in associating each line on the 2 line GS with a different ring tone. The caller ID already tells you who is calling). However, I can put anything I want into the text boxes and nothing happens - I always get the system ring tone. And, what are those stupid little radio boxes for. No matter which one I check, when the screen refreshes it defaults back to the System Ring Tone. Here's what they look like (the o's are supposed to be radio boxes): o System Ring Tone o Custom Ring tone 1, used if incoming caller ID is (Text Box) o Custom Ring tone 2, etc. For me when I select Custom Ring tone 1 I get the first ring tone (what is some stupid piano playing and is pretty much useless) As for ring tone 2 is some piano too. (even more useless now) and ring tone 3 guess piano. (That killed the sense of that function for me) My boss was asking to have have a different ringtone so he can figure out if it's his phone ringing or the one in the office next door. Well I guess with piano's they are the same again. Doooh What's this supposed to mean? It implies that if I select one of the custom ring tones, then the phone will ring on the matching CID, otherwise, it won't ring at all! This feature really needs work. I hope it doesn't wind up like the useless Daylight Savings Time option, which you may have noticed does not pay any attention to the date so you have to log into each and every phone and change the option anyway (please, correct me if this has been fixed). Why bother? I can just as easily change the time zone and get the same effect. GS is obviously targeting their phones for the consumer market where a customer has 1, maybe 2 phones and this kind of thing is irrelevant. The whole concept of their overpriced provisioning system is rather a funny joke in this context, and a rather pretentious one at that. Might be able to hack up some script that changes the function in the tftp file that get's uploaded when the phone is turned on. What would mean everytime there is a timezone change have to run the script and reboot the phones. Ah well we just have 6 of those phones here and so far they are kinda okay (besides mine crashing all the time especially after calls). -- Thomas ___ 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: Grandstream v1.0.4.68 firmware
Stephen R. Besch wrote: Thomas Gallaway wrote: Brian Capouch wrote: Thomas Gallaway wrote: My ringtones just work on all the grandstream's :-) Do the URLS for the ringtones at the top show up as something other than all zeroes? I've fiddled with this until blue in the face, and the ring sounds just like the ring it had before. This is with 1.0.4.68, but it was no different with the earlier supposedly ringtone enabled version. *Product Model: * BT100 *Software Version: * Program--1.0.4.68 Bootloader--1.0.0.16 HTML--1.0.0.31VOC--1.0.0.5 *Custom Ring Tone: * ring1--1.0.0.0 ring2--1.0.0.0 ring3--1.0.0.0 (all zeroes means unavailable or unsupported) -- Thomas Well in my case, the ring versions are all 0.0.0.0 no matter what I do. Could you also post the exact spelling of the binarys on the tftp, including capitalization, access rights and access mode. Mine are ring1.bin, ring2.bin, ring3.bin, all lower case, owned by root, in the asterisk group and with r/w mode=rw_rw_r__ (664), which is the same as all the other items on the server, which do in fact get loaded. Also, I've checked the binaries and they do in fact have version 1.0.0.0 embedded in the file (look at hex offset 6, which is where the version signature of all the GS bianries is located). I also can connect to the tftp server from another machine and successfully get ring1.bin. Perhaps the binaries that came with my copy of the firmware are corrupted. Maybe you could zip up the tones you are using and post them to me so I could see if these fix the problem. I right now run solarwinds tftp server on a winblooze 2000 server. Maybe that's the problem. I had some issues with tftpd on linux. Well actually I just had not the time to mess arround with them hehe. http://atom.port11.net/data/110468.zip (this is the archive I am using) -- Thomas ___ 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
