Re: [Asterisk-Users] Park a call then hunt for a *willing* person
Use the macro feature in dial (CVS-HEAD only, or apply the patch) documented here: http://www.voip-info.org/wiki-asterisk+cmd+dial On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote: Dear All, I like to implement something that does the following: - a call comes in - answered: Please enter your code - caller types a number, eg. '123' - caller hears: we will try to connect you followed by music. - asterisk tries calling a series of predefined numbers, asking each will you accept a caller using code '123', press 1 for yes, 2 for no - when someone accepts, it connects the two callers. Apart from the confirmation message, queueing does this (if I create once queue per allowed 'code'). I have tried using parking, but it does not seem to be possible (at least because we can't get the parked extension number for use in a dialplan). Any suggestions would be appreciated. Thanks, Philip Warner Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Park a call then hunt for a *willing* person
At 04:19 PM 17/04/2005, C F wrote: Use the macro feature in dial (CVS-HEAD only, or apply the patch) documented here: I can't see a way to get Queue to use the macro; it has a limited number of options available. I have tried using 'Local/[EMAIL PROTECTED]' as a queue member, but this seems to fork off a call *and* continue with the call in the queue. Very weird results. http://www.voip-info.org/wiki-asterisk+cmd+dial On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote: Dear All, I like to implement something that does the following: - a call comes in - answered: Please enter your code - caller types a number, eg. '123' - caller hears: we will try to connect you followed by music. - asterisk tries calling a series of predefined numbers, asking each will you accept a caller using code '123', press 1 for yes, 2 for no - when someone accepts, it connects the two callers. Apart from the confirmation message, queueing does this (if I create once queue per allowed 'code'). I have tried using parking, but it does not seem to be possible (at least because we can't get the parked extension number for use in a dialplan). Any suggestions would be appreciated. Thanks, Philip Warner Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Park a call then hunt for a *willing* person
Don't use it with queuing, use it with dial On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote: At 04:19 PM 17/04/2005, C F wrote: Use the macro feature in dial (CVS-HEAD only, or apply the patch) documented here: I can't see a way to get Queue to use the macro; it has a limited number of options available. I have tried using 'Local/[EMAIL PROTECTED]' as a queue member, but this seems to fork off a call *and* continue with the call in the queue. Very weird results. http://www.voip-info.org/wiki-asterisk+cmd+dial On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote: Dear All, I like to implement something that does the following: - a call comes in - answered: Please enter your code - caller types a number, eg. '123' - caller hears: we will try to connect you followed by music. - asterisk tries calling a series of predefined numbers, asking each will you accept a caller using code '123', press 1 for yes, 2 for no - when someone accepts, it connects the two callers. Apart from the confirmation message, queueing does this (if I create once queue per allowed 'code'). I have tried using parking, but it does not seem to be possible (at least because we can't get the parked extension number for use in a dialplan). Any suggestions would be appreciated. Thanks, Philip Warner Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT VoIP related jobs in Eu
I'm posting this here because I'm betting many of you are qualified and someone may be interested. Please, no flames, just act if this is something that interests you, it may be worthwhile. If not move on. Saw this on comp.dcom.voice-over-ip. I want and looked at the site and they do have several ads for voip related positions in at least France and the UK and they seem to work all over Europe. On the other hand, the site doesn't work well in Firefox. Tél: +33153096161 www.clementine-international.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'
I still cannot find it: What does it mean, and how can I fix it? Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:24 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:27 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:28 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' Apr 8 23:50:29 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning, flexible rate not heavily tested!
Any idea? -- SIP Seeding peers from Astdb: '3366' at [EMAIL PROTECTED]:64440 for 3600 -- Saved useragent Sipcom/ATA2000-1.6.11 for peer 3366 -- SIP Seeding peers from Astdb: '886229421761' at [EMAIL PROTECTED]:5060 for 3600 -- Saved useragent Grandstream BT100 1.0.5.18 for peer 886229421761 Ouch ... error while writing audio data: : Broken pipe Warning, flexible rate not heavily tested! Segmentation fault (core dumped) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPswitch: How to use speed dialing?
I tried many different possible ways to us speed dialing, however, I end up in the default context, where the number does not match anything, ... with the result Playing 'demo-congrats' I also could not figure out how to use the tabs Queues and Agents I have not found a new version over the last two days, ... is the author on vacation already ?? Hehehehehe bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting shorter hangup tone (UK)
Hello everybody, I recently got a new phone line from Bulldog-CW (UK). Needless to say that I have connected phone line to my Asterisk system. All seems ok but it does not detect hangups. When the caller hangs-up, the Bulldog line gives a continues tone for a few seconds and then it goes silent. I think that the problem is that the tone does not stay on for long enough (compared to say the BT tone) Does anybody now which settings I need to use/change for a X100P to detect it correctly? For info my system uses 3 X100Ps and the one connected to a BT(UK) line correctly detects CLI and Hangups (using the usual UK patches) and the one connected to the Bulldog line detects CLI correctly but it gets confused with the Hangupi.e. does not detect it! :-) Best regards Vassilis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Point-to-Point Asterisk Link to Reduce Bandwidth
Hi: I want to use G729 codec from my iax connection to my voip provider and later between my two asterisk boxes. G729 bandwidth requirement is relatively low and I intend to reduce more by applying point-to-point-link. While trying to do my homework and figure out how to do that, I appreciate any help in answering my questions. 1- Does point-to-point link requires configuration on both ends (my asterisk and the voip provider). 2- I am behind the nat, does it make any difference or I have to be seen from the internet. Thanks in advance. __ Do you Yahoo!? Plan great trips with Yahoo! Travel: Now over 17,000 guides! http://travel.yahoo.com/p-travelguide ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line name same as user name
Hi, I managed to set up two CISCO 7940 phone with a SIP firmware 7.0, with Asterisk. At the beginning though, I coulnd't understand why they wouldn't work, even if I followed all the instructions found on voip-info.org. Eventually, after some debug and with someone else help, I managed to make them work. All I had to do was to change, in the SIPmac.cnf file, the line_name field to the name of the user_name field. I'm using only one line, but it works. Now my question is, is this the right way to set up all phones or is it a some kind of bug? Thanks in advance. Salvo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPswitch: How to use speed dialing?
Hi Ronald, You posted he same question yesterday and I answered you. Do you till have problems? Thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] I tried many different possible ways to us speed dialing, however, I end up in the default context, where the number does not match anything, ... with the result Playing 'demo-congrats' I also could not figure out how to use the tabs Queues and Agents I have not found a new version over the last two days, ... is the author on vacation already ?? Hehehehehe bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line name same as user name
Hi, I managed to set up two CISCO 7940 phone with a SIP firmware 7.0, with Asterisk. At the beginning though, I coulnd't understand why they wouldn't work, even if I followed all the instructions found on voip-info.org. Eventually, after some debug and with someone else help, I managed to make them work. All I had to do was to change, in the SIPmac.cnf file, the line_name field to the name of the user_name field. I'm using only one line, but it works. Now my question is, is this the right way to set up all phones or is it a some kind of bug? Thanks in advance. Salvo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Illegal instruction (core dumped)
Hi Grabbed the most recent stable asterisk from CVS as documented here: http://www.asterisk.org/index.php?menu=download Didn't bother with zaptel or libpri as I have no Digium hardware nor T1 or E1. Did make install asterisk; make samples. Started asterisk with asterisk -c and it crashes: . . . Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found Illegal instruction (core dumped) Build environment is Mandrake 10.1 official. Didn't have this problem on a Mandrake 10.1 Community box running in vmware - it worked perfectly the first time. Putting noload = res_adsi.so in extensions.conf just causes it to crash elsewhere during the load. Compilation worked fine except for this lot which came out of stderr. Is this normal? In file included from editline.c:18: term.c: In function `term_move_to_line': term.c:556: warning: implicit declaration of function `tputs' term.c:556: warning: implicit declaration of function `tgoto' term.c: In function `term_set': term.c:913: warning: implicit declaration of function `tgetent' term.c:931: warning: implicit declaration of function `tgetflag' term.c:940: warning: implicit declaration of function `tgetnum' term.c:943: warning: implicit declaration of function `tgetstr' term.c:943: warning: passing arg 3 of `term_alloc' makes pointer from integer without a cast In file included from editline.c:18: term.c: In function `term_echotc': term.c:1441: warning: assignment makes pointer from integer without a cast ar: creating libtime.a frame.c: In function `ast_fr_fdread': frame.c:360: warning: assignment discards qualifiers from pointer target type chan_modem_aopen.c: In function `aopen_read': chan_modem_aopen.c:327: warning: assignment discards qualifiers from pointer target type chan_modem_bestdata.c: In function `bestdata_read': chan_modem_bestdata.c:375: warning: assignment discards qualifiers from pointer target type chan_modem_i4l.c: In function `i4l_read': chan_modem_i4l.c:446: warning: assignment discards qualifiers from pointer target type chan_iax2.c: In function `__send_command': chan_iax2.c:3574: warning: assignment discards qualifiers from pointer target type app_mp3.c: In function `mp3_exec': app_mp3.c:169: warning: assignment discards qualifiers from pointer target type app_festival.c: In function `send_waveform_to_channel': app_festival.c:213: warning: assignment discards qualifiers from pointer target type app_nbscat.c: In function `NBScat_exec': app_nbscat.c:147: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `lintoilbc_sample': codec_ilbc.c:95: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `ilbctolin_sample': codec_ilbc.c:110: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `ilbctolin_frameout': codec_ilbc.c:128: warning: assignment discards qualifiers from pointer target type codec_ilbc.c: In function `lintoilbc_frameout': codec_ilbc.c:189: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `lintogsm_sample': codec_gsm.c:85: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `gsmtolin_sample': codec_gsm.c:100: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `gsmtolin_frameout': codec_gsm.c:118: warning: assignment discards qualifiers from pointer target type codec_gsm.c: In function `lintogsm_frameout': codec_gsm.c:203: warning: assignment discards qualifiers from pointer target type src/decode.c: In function `Postprocessing': src/decode.c:25: warning: unused variable `ltmp' src/long_term.c: In function `Long_term_analysis_filtering': src/long_term.c:855: warning: unused variable `ltmp' src/long_term.c: In function `Gsm_Long_Term_Synthesis_Filtering': src/long_term.c:924: warning: unused variable `ltmp' src/lpc.c: In function `Reflection_coefficients': src/lpc.c:214: warning: unused variable `ltmp' src/lpc.c: In function `Quantization_and_coding': src/lpc.c:322: warning: unused variable `ltmp' src/preprocess.c: In function `Gsm_Preprocess': src/preprocess.c:89: warning: unused variable `lsp' src/preprocess.c:49: warning: unused variable `ltmp' src/preprocess.c:50: warning: unused variable `utmp' src/rpe.c: In function `APCM_inverse_quantization': src/rpe.c:365: warning: unused variable `ltmp'
Re: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
tgj wrote: Hi Ronald, You posted he same question yesterday and I answered you. Do you till have problems? Thank you for posting yesterday that there is a new version available, I still have the same problem, I cannot get it to work Thank you that your reply now includes a working example, ... Thank you again! bye Ronald Thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] I tried many different possible ways to us speed dialing, however, I end up in the default context, where the number does not match anything, ... with the result Playing 'demo-congrats' I also could not figure out how to use the tabs Queues and Agents I have not found a new version over the last two days, ... is the author on vacation already ?? Hehehehehe bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPSwitchBoard Version 0.91 Released
Version 0.91 - 17. April 2005. * IPS is now using the context configured in Asterisk for peers - the context on the configuration page is used for Speed Dial Numbers only * New tab page for Speed Dial Numbers Download here: http://ipswitchboard.thorben.dk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPswitch: How to use speed dialing?
Have you tried to change the Context on the configurations page? thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] tgj wrote: Hi Ronald, You posted he same question yesterday and I answered you. Do you till have problems? Thank you for posting yesterday that there is a new version available, I still have the same problem, I cannot get it to work Thank you that your reply now includes a working example, ... Thank you again! bye Ronald Thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] I tried many different possible ways to us speed dialing, however, I end up in the default context, where the number does not match anything, ... with the result Playing 'demo-congrats' I also could not figure out how to use the tabs Queues and Agents I have not found a new version over the last two days, ... is the author on vacation already ?? Hehehehehe bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco/Asterisk codec negotiation problems
On more testing, I conclude that Asterisk isn't being very clever about codec negotiation. My understanding (possibly faulty) from experiments is this. If I have: UA1 -- Asterisk -- UA2 and have disallow/allow entries in UA1's stanza in sip.conf, it seems that the first entry in the allow list is all that's used to choose the codec from UA1. Entries in UA2's stanza and SIP responses from UA2 are not used. If it turns out that UA2 can't support the codec that Asterisk chose for UA1, Asterisk attempts a translation. This occurs even if UA1 and UA2 have a supported codec in common which isn't the one Asterisk chose. If my understanding is correct, this is very inefficient. Worse, if one of the codecs is one it doesn't understand, such as G.729 (without chan_g729a.so) or G.723.1, Asterisk drops the call, even though it could have done pass through. Is my understanding correct? Is this a weakness in Asterisk? Am I missing something elementary? -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ Alistair Cunningham wrote: All, I'm working on an Asterisk 1.0.7 system that is acting as a B2BUA SIP gateway. canreinvite=no is set in the global section of sip.conf, and it's important that it be there. I have Cisco --- Asterisk --- Multiple destinations Some destinations support both G711 and G729, but some only support G729, and some do not support G729. On Cisco, I have: voice class codec 3 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 On Asterisk, I have (irrelevant parts snipped) in sip.conf: [g729only] type = friend host = 192.168.1.1 disallow = all allow = g729 [g711only] type = friend host = 192.168.1.2 disallow = all allow = alaw allow = ulaw [cisco] type = friend host = 10.1.1.1 disallow = all allow = alaw allow = ulaw allow = g729 I've also tried without the disallow and allows in [cisco], and with them in [general]. When I call from the Cisco to g711only via Asterisk, the call works. When I call from the Cisco to g729only via Asterisk, I get: Apr 16 15:56:41 NOTICE[11297]: Unable to find a path from g729 to alaw Apr 16 15:56:41 NOTICE[11297]: Unable to find a path from alaw to g729 In a SIP trace, in the INVITE from Cisco to Asterisk, I see: a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. In the INVITE from Asterisk to g729only: a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. In the 200 OK from g729only: a=rtpmap:18 G729/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. When Asterisk gets this message, it replies to g729only with a ACK then a BYE, and sends a 403 Forbidden to the Cisco. At no point does Asterisk send any message to the Cisco suggesting that Alaw is not acceptable or that G729 is allowed. My question to you, Asterisk-users, is why Asterisk drops the call when both sides have offered G729? It seems to think that the Cisco is Alaw only; this is the default codec on Cisco, but it is offering other codecs as well. I've tried various settings. I can also make G729 work but G711 fail, but can't do both at the same time. I've also tried the following on Cisco: voice class codec 2 codec preference 1 g729r8 codec preference 2 g729br8 codec preference 3 g723r63 codec preference 4 g723r53 codec preference 5 g723ar63 codec preference 6 g723ar53 codec preference 9 g711alaw codec preference 10 g711ulaw codec preference 11 gsmfr codec preference 12 gsmefr With the dial-peer changed to use it, but this doesn't help either. (Hostnames and IP addresses changed to protect the guilty, i.e. me) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: IPswitch: How to use speed dialing?
tgj wrote: Have you tried to change the Context on the configurations page? yes, .. please read below please post an EXAMPLE how you think it works. thank you bye Ronald thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] tgj wrote: Hi Ronald, You posted he same question yesterday and I answered you. Do you till have problems? Thank you for posting yesterday that there is a new version available, I still have the same problem, I cannot get it to work Thank you that your reply now includes a working example, ... Thank you again! bye Ronald Thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] I tried many different possible ways to us speed dialing, however, I end up in the default context, where the number does not match anything, ... with the result Playing 'demo-congrats' I also could not figure out how to use the tabs Queues and Agents I have not found a new version over the last two days, ... is the author on vacation already ?? Hehehehehe bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel fxo late distinctive ring
I have a TDM400 with an fxo card installed; zaptel.conf is setup for distinctive ring, but it only partly works: Our distinctive rings have the same sound in the very first ring (one long tone). After that, they differ. As a result, all calls produce a dring signature of 0,0,0. If I setup the dialplan to immediately hangup, then zaptel hears the next ring and reports the different signatures correctly. This is not ideal for three reasons: (a) the caller id is part of the first ring, so it is lost and (b) it means a lot of rings occur before the phone is answered and (c) it is inelegant. Does anyone know of parameters I can set within the zaptel driver or config files to cause it to wait a little longer before reporting the dring type? Or any other solution... Thanks again, Philip Warner Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom and hint priority
Hello, I had the same problem. I solved it by putting the context of the phone in sip.phone as the same context where the hint statement is: i.e.: sip.conf [1713] context=phones extensions.conf [phones] ;1713 exten = 1713,hint,sip/1713 exten = 1713,1,Playback(transfer,skip) ; Please hold while... exten = 1713,2,Macro(stdexten,1713,sip/1713) hope it will work for you too - Original Message - From: Lance Grover [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 17, 2005 6:23 AM Subject: Re: [Asterisk-Users] snom and hint priority I have set up Hint on all my extensions, according to all I have found out, the correct way, however I do not get anything on the phone. Is there something I am missing? I have one of these snom 220's with the side car, and another 220. I am running an RPM version of asterisk and have also tried this on a compiled version of asterisk from the CVS tree. Neither way did it work, is there some thing else I am missing? I set it up as Destination with the sip URL of the extension and my dial plan looks like this: ;1713 exten = 1713,hint,sip/1713 exten = 1713,1,Playback(transfer,skip) ; Please hold while... exten = 1713,2,Macro(stdexten,1713,sip/1713) as you can see I use a Macro but I do not try to put the hint in the Macro, also I have tried this without the Macro. I have rebooted the phone and restarted asterisk after each change. Can someone please help me out? Thanks a ton, -Lance On 4/13/05, Josh Dady [EMAIL PROTECTED] wrote: (boy mail in this list piles up fast when I can't check it) On Apr 8, 2005, at 10:03 AM, Michael George wrote: - It appears that the extension used with the hint must be the same as the extension used to dial that channel. So if extension 22 will ring Zap/2, then exten = 22,hint,Zap/2 will work, but exten = 222,hint,Zap/2 will not. Why is that? The extension is how asterisk maps SIP URLs to chunks of your dialplan -- if you program a button on a snom to dest sip:[EMAIL PROTECTED], the phone will use that same URL for both dialing and subscribing to extension state. Unless you have a phone that lets you specify different URLs for dialing and subscribing to state, they have to match in asterisk. - If I am correct in the above, then there is no way for me to monitor a channel that is not an extension. As an example, I have a TDM400 with 3 FXS (Zap/1-3 on extensions 21-23) and 1 FXO (Zap/4) as well as a VoIP channel for dialing out. I can monitor the states of the extensions with extension entries like exten = 21,hint,Zap/1 but I cannot monitor the state of the FXO with exten = 0,hint,Zap/4 because 0 is not the extension of Zap/4. Indeed, Zap/4 has no extension. Is it not possible to monitor that line, then? There has to be a SIP URL for the phone to subscribe to -- if you put: exten = zap4,hint,Zap/4 in your extensions.conf (with no zap4,1,... entry) it wouldn't be dialable (although the phone would still try if you pushed it) but would have a valid SIP URL. -- Joshua P. Dady ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Lance Grover ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPSwitchBoard Version 0.91 Released
Thorben Jensen wrote: Version 0.91 - 17. April 2005. * IPS is now using the context configured in Asterisk for peers - the context on the configuration page is used for Speed Dial Numbers only * New tab page for Speed Dial Numbers Download here: http://ipswitchboard.thorben.dk Thorben, I hope you find some time to make all more smoothly. It is a great product, but there are still some unclear things. Following problems I have encountered: 1. The help system is still in the very first stage, ... a typical engineer habit ;-) I hope you can add in the next version a little bit how to use IPSwitchBoard 2. The Zap Extension will pop up all the time in Main Extension tab, even you delete it already and / or renamed it to another tab 3. One IAX2 is simple to taken The three lines in Exensions / Extensions tab look like: IAX2 623 IAXy at home 623 Unspecified Internal (it is in the moment not connected) IAX2 NuFoneNuFone (Toll free USA) 6.225.202.72 Lines IAX2 demoDigium16.207.245.47 Main Extensions The button NuFone is always EMTPY (in Panel / Lines) 4. I cannot find out the purpose of Shared Extensions File in Config 5. Speed Dials I cannot get it to work, ... Maybe we misunderstand the purpose, please correct me, if I am wrong. I think it should let me key in a Name (Peter) and a Caller ID (phone number of him). I tried with [EMAIL PROTECTED], I tried 901, . It always tells me that I used the demo of asterisk!!! 6. Queues and Agents I have not setup yet at Asterisk, of course I cannot get here anything too. 7. Calls I am not sure what the purpose is of it. Maybe the last 50 Incoming / Outgoing calls, or forever Depending on the answer before, it would be nice to have a search for some cases. It has also a sorting problem of one digit hours and two digit hours 10 pm is listed before 5 pm 8. Installation It would be nice if it ask automatically to uninstall the previous version while it installs the new one, instead to go the extra mile via control panel to uninstall the previous version. All in all, I think it will soon a good product. I admire your effort to upgrade all the time,... bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPswitch: How to use speed dialing?
Hi Ronald, I must admit I am getting confused now. I understand that you have a problem getting Speed Dial Buttons to work. The problem as I understand it is that the calls are placed in the wrong context. To solve that problem I have asked you to make sure that you have typed a valid context on the configuration page. Have you tried that? I think thats all you need to do, how do I post an example of that? It's a fairly easy thing to do. Thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] tgj wrote: Have you tried to change the Context on the configurations page? yes, .. please read below please post an EXAMPLE how you think it works. thank you bye Ronald thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] tgj wrote: Hi Ronald, You posted he same question yesterday and I answered you. Do you till have problems? Thank you for posting yesterday that there is a new version available, I still have the same problem, I cannot get it to work Thank you that your reply now includes a working example, ... Thank you again! bye Ronald Thorben Ronald Wiplinger [EMAIL PROTECTED] skrev i en meddelelse news:[EMAIL PROTECTED] I tried many different possible ways to us speed dialing, however, I end up in the default context, where the number does not match anything, ... with the result Playing 'demo-congrats' I also could not figure out how to use the tabs Queues and Agents I have not found a new version over the last two days, ... is the author on vacation already ?? Hehehehehe bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IPSwitchBoard Version 0.91 Released
Thorben, I hope you find some time to make all more smoothly. It is a great product, but there are still some unclear things. Following problems I have encountered: 1. The help system is still in the very first stage, ... a typical engineer habit ;-) I hope you can add in the next version a little bit how to use IPSwitchBoard 2. The Zap Extension will pop up all the time in Main Extension tab, even you delete it already and / or renamed it to another tab 3. One IAX2 is simple to taken The three lines in Exensions / Extensions tab look like: IAX2 623 IAXy at home 623 Unspecified Internal (it is in the moment not connected) IAX2 NuFoneNuFone (Toll free USA) 6.225.202.72 Lines IAX2 demoDigium16.207.245.47 Main Extensions The button NuFone is always EMTPY (in Panel / Lines) 4. I cannot find out the purpose of Shared Extensions File in Config 5. Speed Dials I cannot get it to work, ... Maybe we misunderstand the purpose, please correct me, if I am wrong. I think it should let me key in a Name (Peter) and a Caller ID (phone number of him). I tried with [EMAIL PROTECTED], I tried 901, . It always tells me that I used the demo of asterisk!!! 6. Queues and Agents I have not setup yet at Asterisk, of course I cannot get here anything too. 7. Calls I am not sure what the purpose is of it. Maybe the last 50 Incoming / Outgoing calls, or forever Depending on the answer before, it would be nice to have a search for some cases. It has also a sorting problem of one digit hours and two digit hours 10 pm is listed before 5 pm 8. Installation It would be nice if it ask automatically to uninstall the previous version while it installs the new one, instead to go the extra mile via control panel to uninstall the previous version. All in all, I think it will soon a good product. I admire your effort to upgrade all the time,... Hi Ronald, I will try to answer all your questions: 1. The help system is not up-to-date - The release notes on the web site are. 2. That's seems like a bug, I will investigate. 3. Have you got a button on the panel? (But with no text on it)? 4. The shared extension file can be used if you place an XML file on a network drive, and all client point to that file. The file will be read every time IPS is started. It's a good way to share extensions among a lot of users 5. I misunderstood you. I see the problem; you have to type the number in the name column and the name in the callerid column (I admit that that is not clear). 6. The Queues/agents pages just show the status of your Asterisk configured Queues/Agents 7. Calls are logging your incoming and outgoing calls (forever) you can delete if the list gets to long. I will look at the sorting issue. 8. I am using Whidbey.NET and I am still to work out how that is done, I know it's very annoying to have to remove before you can install. I wan to thank you for all your comment; it's very useful for me. Regards Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bridging 2 Zap channels
On Saturday 16 April 2005 14:00, [EMAIL PROTECTED] wrote: On Fri, 15 Apr 2005, Paul Hewlett wrote: I am running * 1.0.6 with 8 analogue phone lines connected to 2 cards - lspci reveals these as : The problem is that under certain circumstances (which I am unable to determine) * bridges 2 of the Zap channels together even though I can see no possible way in the dialplan. This then permanently consumes 2 lines leaving only one available. I have been watching the system for 2 days now and have managed to trap it into this condition twice - the system is only under light load. Hi Paul, Hi Steve I believe I'm coming to you on Tuesday to look at this and some other things. But I think I know what's wrong from your description... Look forward to seeing you.. Are you using SNOM phones? Go to the advanced setup and turn off the bridge calls on hangup option. Yes we are using SNOM phones but they do not have the bridge calls on hangup option on the advanced setup page Paul -- Paul Hewlett (Linux #359543) Tel: +27 21 852 8812 Cel: +27 72 719 2725 Fax: +27 86 672 0563 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_perl compile problem
Hi ALL; Itriedto compile res_perl module with Asterisk, but It failed.I use both Asterisk and Asterisk-addons lates CVShead. I did as follows: 1- make a patch to Asterisk Makefile. 2- Try to re-build Asterisk. BUTit says: gcc -c perlxsi.c -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm -I/usr/lib/perl5/5.8.0/i386-linux-thread-multi/CORE -o perlxsi.ogcc: perlxsi.c: No such file or directorygcc: no input filesmake: *** [perlxsi.o] Error 1 How can I get the file "Perlxsi.c"???/ Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco mgcp and CARD.XML
Is there a working CARD.XML for cisco MGCP phones? The one on the cisco site is old and it's not working with the new firmwares. Thx Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] first few seconds of outgoing calls cut off
On Sat, 2005-04-16 at 13:50 -0700, snacktime wrote: This also happens to me when I call into my own * box voice system unless I'm very careful about adding appropriate wait statements after answering the line. Not sure if this is related to the above problem, but it made me wonder if an * box somewhere in the path of my outgoing calls might be the culprit. Any thoughts? This problem could be caused by a SPF firewall somewhere in the path. I had a similar problem where the firewall was dropping RTP packets in one direction until it saw a packet in the other direction. Removing the stateful firewall rules and replacing them with pairs of non stateful rules fixed the problem. Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *8 nor *8# works for me!
On Fri, Apr 15, 2005 at 10:39:44PM +0800, Ronald Wiplinger said: Eric Wieling wrote: I have put into each phone settings (sip.conf and zapata.conf) in my office: callgroup=1 pickupgroup=1 I cannot pickup any calls from another phone!! What do I miss here? Your SIP phone is eating the *8. You need to look at your SIP phone docs, not Asterisk What am I going to look for, e.g., in a manual for snom 190 and a Budgetone ??? See the Wiki: http://www.voip-info.org/wiki-Asterisk+config+features.conf I had the same problem with Cisco ATA's screwing with the *, so I changed mine to a normal number and everything works great. I never did figure out how to make the cisco pass the *8 properly. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Park a call then hunt for a *willing* person
At 05:02 PM 17/04/2005, C F wrote: Don't use it with queuing, use it with dial One problem with this: queueing gives a context menu. Just using a series of 'Dial' commands means that I lose the ability for the caller to have a context menu without putting a call to WaitExten or Background (both of which pause MoH). On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote: At 04:19 PM 17/04/2005, C F wrote: Use the macro feature in dial (CVS-HEAD only, or apply the patch) documented here: I can't see a way to get Queue to use the macro; it has a limited number of options available. I have tried using 'Local/[EMAIL PROTECTED]' as a queue member, but this seems to fork off a call *and* continue with the call in the queue. Very weird results. http://www.voip-info.org/wiki-asterisk+cmd+dial On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote: Dear All, I like to implement something that does the following: - a call comes in - answered: Please enter your code - caller types a number, eg. '123' - caller hears: we will try to connect you followed by music. - asterisk tries calling a series of predefined numbers, asking each will you accept a caller using code '123', press 1 for yes, 2 for no - when someone accepts, it connects the two callers. Apart from the confirmation message, queueing does this (if I create once queue per allowed 'code'). I have tried using parking, but it does not seem to be possible (at least because we can't get the parked extension number for use in a dialplan). Any suggestions would be appreciated. Thanks, Philip Warner Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Illegal instruction (core dumped)
On April 17, 2005 05:55 am, Tom Fanning wrote: Illegal instruction (core dumped) Sounds like you have compiled asterisk for a processor that is greater than the processor you're running on. I.e. compiled and told it to use P4 instructions when you're on a P3, or maybe even told it to use MMX on a Via processor... -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_dtmftotext.c
Hi, I was looking for a way to pass alphanumeric variables to asterisk via the keypad, found this application app_dtmftotext.c and its use instructions on the wiki, but with no compiling/installation instructions. Can anybody be of help here? Thx Ezabi signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Park a call then hunt for a *willing* person
FWIW, I finally got it going using queues. The queue has one member which is a Local/[EMAIL PROTECTED] number. The context it points to uses Dial with the M() option, and it all seems to work...MoH runs all the time, and the caller can leave the queue via voicemail. Thanks for the help etc. At 04:19 PM 17/04/2005, C F wrote: Use the macro feature in dial (CVS-HEAD only, or apply the patch) documented here: http://www.voip-info.org/wiki-asterisk+cmd+dial On 4/17/05, Philip Warner [EMAIL PROTECTED] wrote: Dear All, I like to implement something that does the following: - a call comes in - answered: Please enter your code - caller types a number, eg. '123' - caller hears: we will try to connect you followed by music. - asterisk tries calling a series of predefined numbers, asking each will you accept a caller using code '123', press 1 for yes, 2 for no - when someone accepts, it connects the two callers. Apart from the confirmation message, queueing does this (if I create once queue per allowed 'code'). I have tried using parking, but it does not seem to be possible (at least because we can't get the parked extension number for use in a dialplan). Any suggestions would be appreciated. Thanks, Philip Warner Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Philip Warner| __---_ Albatross Consulting Pty. Ltd. |/ - \ (A.B.N. 75 008 659 498) | /(@) __---_ Tel: (+61) 0500 83 82 81 | _ \ Fax: (+61) 03 5330 3172 | ___ | Http://www.rhyme.com.au |/ \| |---- PGP key available upon request, | / and from pgp.mit.edu:11371 |/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] new install
With the 2.6 kernel, you can just load ztdummy and not worry about the USB controller. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 190: Unknown SIP command 'PUBLISH'
I still cannot find it: What does it mean, and how can I fix it? Apr 8 23:50:23 NOTICE[12235]: chan_sip.c:8486 handle_request: Unknown SIP command 'PUBLISH' from '192.168.250.108' I think your phone is trying to do overlap dialing but Asterisk does not support this yet. I don't think there is any way to turn it off. Just ignore it. The phones will still work. -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP signature.asc Description: Digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Illegal instruction (core dumped)
[EMAIL PROTECTED] wrote on 04/17/2005 11:02:51 AM: On April 17, 2005 05:55 am, Tom Fanning wrote: Illegal instruction (core dumped) Sounds like you have compiled asterisk for a processor that is greater than the processor you're running on. I.e. compiled and told it to use P4 instructions when you're on a P3, or maybe even told it to use MMX on a Via processor... This is especially true for Via processors. They identify themselves as 686 processors, but do not implement the CMOV instruction, which GCC considers to be a 686-class instruction. Do a search for Via CMOV Linux compile or somesuch on Google and you will see the modifications you will need to make to the makefile to address this. Incidentally, I believe that the latest processors (the Nehemiah C5P found on EPIA MII boards) support CMOV. I'm less sure, but the Nehemiah processors themselves may also support CMOV. The Samuel processors, though, do *not* support CMOV. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk
I'm in the UK so numbers are generally started with a zero. The
dialstring sent to the sipura is fine, running asterisk
-vvvc gives me called number@101.
Where 101 is the extention number of the sipura.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed
Greenberg
Sent: 16 April 2005 14:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk
Hi Razza,
I don't know what country you are in, or what your country's telephone
numbers look like, but it seems from your dialplan that if you dial an
outside number it needs to start with 0X.
So if you dial 012345, the Sipura will dial 012345 on the fxo port.
If your line needs to dial 12345, you should use ${EXTEN:1} to drop the
zero off the beginning.
I recommend that you run the console with verbose on (asterisk -rv)
and
watch to see what is actually being dialed on the Sipura.
Best,
/edg
If that is not the problem,
--On Saturday, April 16, 2005 11:50 AM +0100 Razza
[EMAIL PROTECTED] wrote:
All,
Further to my note below, I now have incoming working - yipee! (and
seem to have identified a problem with the G711A codec in the latest
sipura firmware - although need to do some checking). This box sounds
great compared to the echo ridden FXO and gives me an FXS for very
little more cash.
I now have a really strange issue for outgoing calls, everything seems
ok including the SIP messages (i.e. dialled number@sipura ext) but
I am always getting through to a wrong number (fortunately I'm doing
this on a Sunday and it's a business number so I'm just getting their
answer machine).
I have included excepts from my test extensions.conf and sip.conf
files, could someone please confirm these are ok (for my own sanity)?
The other strange thing is the sipura info tab tells me 'Last Called
PSTN Number' is correct.
I assume I have got something very wrong with the sipura config,
although have not changed anything - so assistance on the sipura would
be greatly appreciated.
-
*** extensions.conf
-
[general]
static=yes
writeprotect=no
[globals]
CC=UK
CONSOLE=Console/dsp
[sip_home]
exten = 100,1,SETCIDNUM(${CALLERIDNUM:1}) ; strips leading character
added to CLI by the SPA3K to frig no answer issue
exten = 100,2,Dial(SIP/budget1,25,tr)
exten = _0X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)
exten = 105,1,Dial(SIP/budget1,20tr)
-
*** sip.conf ***
-
[general]
% -- SNIP --- %
[101]
;PSTN
type=friend
regexten=101
username=983
secret=razza
context=sip_home
port=5080
host=dynamic
nat=no
canreinvite=no
disallow=all
;allow=alaw
allow=ulaw
[budget1]
type=friend
regexten=105
username=budget1
secret=razza
context=sip_home
callerid=Kitchen 105
host=dynamic
nat=no
;canreinvite=no
disallow=all
;allow=alaw
allow=ulaw
Regards,
Ray
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Razza
Sent: 16 April 2005 00:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk
Pete wrote:
The comments about it being an ugly hack arent really correct. The
Sipura is really built for standalone useage wiht a sip provider
however it does work well with asterisk.
Follow this thread
http://voxilla.com/forum-viewtopic-t-1335.html
it works and it works **VERY** well :-)
Pete
Help!!!
I have spent the whole day trying to get this to work and simply cant,
I'm aware the instructions are very simple but there is no sip traffic
generated to the * server from the SPA-3000 when I call my PSTN number
(outgoing from sip phone to spa-3000 through * is fine) - are there
other settings I am missing?
As I am in the UK I have also changed the line impedences according to
http://www.sinet.bt.com/351v4p2.pdf and have changed the 'Caller ID
Method' (in regional tab) to 'ETSI FSK WithPR (UK)' but still nothing.
Anyone able to send me screen dumps of their config or advise?
Ray.
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[Asterisk-Users] AMP + POLYCOM
Is there a Plugin for AMP to ease Polycom 500's Configurations? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_perl compile problem
Hi, gcc -c perlxsi.c -D_REENTRANT -D_GNU_SOURCE -fno-strict-aliasing - D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gdbm - I/usr/lib/perl5/5.8.0/i386-linux-thread-multi/CORE -o perlxsi.o gcc: perlxsi.c: No such file or directory gcc: no input files make: *** [perlxsi.o] Error 1 How can I get the file Perlxsi.c???/ nowhere, since is created automatically from res_perl Makefile. honestly I didn't have to modify * makefile. here I just did make clean ; make ; make install on res_perl dir, and all went ok. matteo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwidth Reduction using Compressed RTP
Hello: I read many documents about reducing the codec bandwidth by 1)compressing the rtp header and 2)implementing point-to-point link. But none of these documents mentioned how to implement it. So I wonder why there is not much resources about something valuable like this which interest many people, or I just don't see it. I am hoping someone can help. this is one of the many resources I read: http://www.newport-networks.com/whitepapers/voip-bandwidth1.html Thanks; __ Do you Yahoo!? Plan great trips with Yahoo! Travel: Now over 17,000 guides! http://travel.yahoo.com/p-travelguide ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPP g729 x86_64
Hi, I 'm using a server DL145 with AMD opteron processors, with TE410P Digium Quad-Span card. The server is running RHEL4 x86_64. And have problem to compile codec g729 from http://www.readytechnology.co.uk/open/g729/, but ipp sample speech code not problem compile with ia32 or em64t. use l_ipp_ia32_itanium_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/ia32_itanium/lib -lippscmerged -lippsrmerged -lippsmerged -lippcore -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Iand use from l_ipp_em64t_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/em64t/lib -lippscem64t -lippsrem64t -lippsem64t -lippcoreem64t -L/opt/intel/ipp41/em64t/sharedlib/linuxem64t -lguide -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Any thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] problem connecting multiple boxes via IAX2
Send me a copy of your iax.conf and your extensions.conf. I'll look at it. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of MobilPete Sent: Saturday, April 16, 2005 6:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problem connecting multiple boxes via IAX2 Senerio multiple * boxes connecting to a central * box with T1 card via IAX2. 1box 1 abd 2 work fine all the time box 3 - after approx 10-15 minutes with no calls - central box with T1 card fails to deliver incoming calls to box 3. Connectivity is good, * exten-2-exten good in order to allow incoming calls again, we only need to make 1 outbound call from box 3. Then everything works well again. Can anyone shine some light on this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk
Greg, Have you checked the 'PSTN Dialing Delay:' setting under the 'PSTN Line' tab, I suggest this is at least 1 (second), just to let things stabalise. % -- SNIP -- % Greg Wrote: Lucky you, my spa-3000 likes to dial 911. So far the local cops have been nice about it though. (my mobile number ends in 9110) % -- SNIP -- % ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
On Sun, 17 Apr 2005 01:39:09 -0400, Karl J. Vesterling wrote: H.323 will not traverse NAT. Sorry... I know, I was a big proponent of it when H.323 was the only standard VoIP protocol out there. Probably because when it came out NAT wasn't even thought of. The problem is that the control channel in H.323 discloses the internal IP address, and the various connections attempt to connect to each other. So you wind up with problems like audio only in one direction, etc... I thought SIP had the same problem though. Can't this be solved with address translation inside asterisk? You know, like the externip, localnet, and nat=yes options in sip.conf? Or is it simply impossible due to limitations within the H.323 spec? It's difficult to find information about this sort of thing on the internet. H.323 is such a broad spec... Although I get get this to solve part of the problem back in year 2K: http://openh323proxy.sourceforge.net/ It never solved the problem entirely, and I had a lot of H.323 equipment at the time, so I was somewhat disappointed when the asterisk project said integration with H.323 was impossible due to licensing issues. (Bummer)... Your best bet is to abandon H.323 and find something other than GnomeMeeting. That is unless you want to carry a portable asterisk box with you... Wait a sec... COME TO THINK OF IT! Why not run asterisk on your linux box that you are running GnomeMeeting on, and use it to convert between H.323 and IAX and SIP??? After all, it is a penguin... That's certainly a good alternative. I'm currently in the process of hacking up the latest linphone (1.0.1) to fix a few personal show-stoppers. If I can get it to the point that I like it, then I'll probably just go with linphone. But you're right. If it's took much work, then I'll probably just start running asterisk on my laptop to do H.323 to SIP conversions. Thanks for the suggestion! I hadn't thought of that yet. I'd been looking at things like the commercial sip323 program, but I hadn't thought of doing it with a local copy of asterisk. -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loop Detection
This is very interesting to me since I am in the process of setting up SER to Asterisk in a similar scenario. I'm surprised there haven't been more posts. Maybe include SER - Asterisk in the title. There are other posters on the list who use SER and Asterisk together who surely must have encountered (overcome?) this problem since it is so fundamental. Perhaps a bug should be raised? Regards Cameron - Original Message - From: Daniel Corbe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 14, 2005 7:29 AM Subject: [Asterisk-Users] Loop Detection Hello, Is there any way to turn Loop Detection off or tune the params a bit? I am having an issue with Call Forwarding on my SIP Proxy Server which is causing me great pains. Here is the issue: 1) I have a SIP UA which registers with a SER proxy server. 2) I have an Asterisk TDM gateway in my network, also which registers with SER 3) A call comes in through the PSTN to the Asterisk Gateway. The Asterisk gateway sends the call to SER destined for my SIP UA 4) SER sees that the SIP UA has call forwarding enabled so it creates a new outbound call with the same Call ID but it has a different TAG= line and Max-Forwards is set to 70. 5) Since the fowarding number is out on the PSTN, SER routes the call back through the same * gateway. 6) Asterisk rejects the phone call with Loop Detected According to my interpretation of the RFC, it is more correct to base loop detection off of the TAG= than it is off of the Call ID. Having said that, SER also sets the Max Forwards on the call. Is there any way at all to get Asterisk to either base its loop detection off the TAG= or respect the Max-Forwards setting? I've also attached a libpcap packet dump of a phone call. 389.764074 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 389.885825 66.165.175.44 - 62.25.108.211 SIP Status: 401 Unauthorized 389.885999 62.25.108.211 - 66.165.175.44 SIP Request: ACK sip:[EMAIL PROTECTED] 389.886104 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 390.145261 66.165.175.44 - 62.25.108.211 SIP Status: 100 trying -- your call is important to us 390.257658 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.257706 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 390.801964 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.802007 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 391.901785 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 391.901829 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 393.991808 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 393.991851 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 401.223872 62.25.108.211 - 66.165.175.44 SIP Request: CANCEL sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 152
On April 17, 2005 05:55 am, Tom Fanning wrote: Illegal instruction (core dumped) Sounds like you have compiled asterisk for a processor that is greater than the processor you're running on. I.e. compiled and told it to use P4 instructions when you're on a P3, or maybe even told it to use MMX on a Via processor... Have just shoved PROC=i586 in the Makefile along with some commenting to see what happens. It's compiling right now. It is indeed on a Via Epia board. Cheers Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
Jesse Guardiani [EMAIL PROTECTED] writes: Wait a sec... COME TO THINK OF IT! Why not run asterisk on your linux box that you are running GnomeMeeting on, and use it to convert between H.323 and IAX and SIP??? After all, it is a penguin... That's certainly a good alternative. I'm currently in the process of hacking up the latest linphone (1.0.1) to fix a few personal show-stoppers. If I can get it to the point that I like it, then I'll probably just go with linphone. But you're right. If it's took much work, then I'll probably just start running asterisk on my laptop to do H.323 to SIP conversions. Thanks for the suggestion! I hadn't thought of that yet. I'd been looking at things like the commercial sip323 program, but I hadn't thought of doing it with a local copy of asterisk. If your only reason to stick to H323 is Gnomemeeting you could try other softphones as well. Especially, the XLite beta for Linux looks promising, and some people like SJphone for Linux. Also, SIP support for Gnomemeeting is underway, but development is slow. I'm constantly pointing out to them how much interest there is, but things still seem to take their time ... Finally, on a recent discussion about the future design of GM on their list, I was surprised to learn that quite a few people really use it for direct PC to PC video calls over the internet. So somehow, after extensive NAT and router fiddling I guess, direct calls apparently work even with H323 (there is already support built into GM for external IP address discovery, as you know, so those remarks about transmission of bogus IP addresses on H323 level probably don't really apply in this case). Anyway, I myself use the setup recommended above, i.e. local * server as protocol translator, and it works reasonably well. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loop Detection
I keep getting the same answer from people Well the SIP implementation is fine if you use XXX IP Phone so obviously Asterisk was never designed to be used as a TDM gateway but merely as a PBX server only. On 4/17/05, Cameron Beattie [EMAIL PROTECTED] wrote: This is very interesting to me since I am in the process of setting up SER to Asterisk in a similar scenario. I'm surprised there haven't been more posts. Maybe include SER - Asterisk in the title. There are other posters on the list who use SER and Asterisk together who surely must have encountered (overcome?) this problem since it is so fundamental. Perhaps a bug should be raised? Regards Cameron - Original Message - From: Daniel Corbe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 14, 2005 7:29 AM Subject: [Asterisk-Users] Loop Detection Hello, Is there any way to turn Loop Detection off or tune the params a bit? I am having an issue with Call Forwarding on my SIP Proxy Server which is causing me great pains. Here is the issue: 1) I have a SIP UA which registers with a SER proxy server. 2) I have an Asterisk TDM gateway in my network, also which registers with SER 3) A call comes in through the PSTN to the Asterisk Gateway. The Asterisk gateway sends the call to SER destined for my SIP UA 4) SER sees that the SIP UA has call forwarding enabled so it creates a new outbound call with the same Call ID but it has a different TAG= line and Max-Forwards is set to 70. 5) Since the fowarding number is out on the PSTN, SER routes the call back through the same * gateway. 6) Asterisk rejects the phone call with Loop Detected According to my interpretation of the RFC, it is more correct to base loop detection off of the TAG= than it is off of the Call ID. Having said that, SER also sets the Max Forwards on the call. Is there any way at all to get Asterisk to either base its loop detection off the TAG= or respect the Max-Forwards setting? I've also attached a libpcap packet dump of a phone call. 389.764074 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 389.885825 66.165.175.44 - 62.25.108.211 SIP Status: 401 Unauthorized 389.885999 62.25.108.211 - 66.165.175.44 SIP Request: ACK sip:[EMAIL PROTECTED] 389.886104 62.25.108.211 - 66.165.175.44 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 390.145261 66.165.175.44 - 62.25.108.211 SIP Status: 100 trying -- your call is important to us 390.257658 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.257706 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 390.801964 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 390.802007 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 391.901785 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 391.901829 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 393.991808 66.165.175.44 - 62.25.108.211 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED]:5060, with session description 393.991851 62.25.108.211 - 66.165.175.44 SIP Status: 482 Loop Detected 401.223872 62.25.108.211 - 66.165.175.44 SIP Request: CANCEL sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?
Hello All, I have been trying a did company for a few days. I find the service decent, but sound quality only moderate. Rather than spending 35 or so for monthly with did, I am considering an isdn bri at this location. How much more stable and reliable is bri or pri versus a voip did service? I like the concept of a bri more, but I do not get cid generation. Would anyone suggest bri over voip where available? I must say, I prefer higher voice quality. If anyone finds bri to be worth it (at about 54/month plus usage) please let me know what you think. Thanks, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP to PTSN provider
I have to agree that voipjet is a good service. If only they had did's it would be even better, but I like the fact that outgoing cid works well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg Sent: Saturday, April 16, 2005 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VOIP to PTSN provider 800 numbers are free to the caller because the recipient pays the charge. Voipjet has no way to get paid anything for carrying the calls, hence they are unwilling to use their resources to move calls with no revenue. Can you blame them? :) /edg --On Saturday, April 16, 2005 9:44 PM +0100 Chris Hills [EMAIL PROTECTED] wrote: Andy Hamilton wrote: I use voipjet and am quite pleased. Good enough rates and no noticeable quality issues. http://www.voipjet.com Plus, you can even test it before you buy. On their pricing page, they have:- There are some providers who can terminate some, but not all, 1800 numbers for free. (If they could terminate all 1800 numbers for free, then we'd use them!) I don't understand - I thought all 1800 numbers were free? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] High Availability - Again
Hello to all. I saw on http://www.intel.com/software/products/cluster/clustertoolkit/features.htm a software (or feature) a Cluster Toolkit for Linux distributions that use Intel Pentium 4 Processors. Does anyone know if is possible to use The \Intel Cluster Software\ for High Availability of Asterisk Systems? For exemple: 1 x * box with 300 IP Phones can be switched to other one (Cluester ou Backup) using this kind of Intel software? Any comments? Alexandre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP to PTSN provider
Ed Greenberg wrote: 800 numbers are free to the caller because the recipient pays the charge. Voipjet has no way to get paid anything for carrying the calls, hence they are unwilling to use their resources to move calls with no revenue. Can you blame them? :) Well it's not a problem, I can terminate calls to 1800 using e164.org for free anyway. It just seems a bit stingy! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800DID's?
Please share with us the name of the company you had a bad experience with. Then we can avoid the same problem. Thanks Cameron - Original Message - From: Linn Boyd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 15, 2005 2:33 AM Subject: [Asterisk-Users] Who is a QUALITY IAX Termination Provider for 800DID's? I have looked for a quality IAX provider for 800 DID's we currently have two, one is ok and the other is just not of quality, but last night we got an email after a complaint of quality earlier in the day and this is what it said. Remember I never did request a network change, but I just wanted my quality fixed, they have all kinds of contact information and they could have let me know outside of voice mail. I have been trying to call them and trying to email them ever since I found out. We have migrated your account to an alternate network. Please change the host you register to, send traffic to and receive traffic from in your appropriate config files to iax01.someprovider.net Well, our phones for which our main customers dial in on are now DOWN and we had a 250k mailer that went out three days ago with the phone number on it. I hate to loose business, but what I hate worse is that I am probably loosing my job over this one. Can anyone give me any information? Cost is not an issue, but uptime and service is! Also we need a true 800 number. -Linn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP to PTSN provider
No, I suppose you can't blame them What is FWD's motivation (or IAXtel, etc) to provide this service, then? -Andy On 4/16/05, Ed Greenberg [EMAIL PROTECTED] wrote: 800 numbers are free to the caller because the recipient pays the charge. Voipjet has no way to get paid anything for carrying the calls, hence they are unwilling to use their resources to move calls with no revenue. Can you blame them? :) /edg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp and cvs head
Guys. Ive done some searching and seems that installing spandsp on cvs head had been a pain because of changes on the patch. Anybody has a howto on installing spandsp on the recent cvs head? And how they got receiving and sending faxes worked out? Hope you can help. Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848
Just curious what syntax did you use to load the VMware tools on Fedora Core 3? Thanks, Sean On Sat, 16 Apr 2005 16:50:56 +0200, [EMAIL PROTECTED] wrote: I installed asterisk 1.0.7 successfully on VMware workstation with fedora 3 as guest. Of course without any hardware only pure asterisk. It works fine for testing. SCollins wrote: Newbie Question Has anybody installed [EMAIL PROTECTED] on VMware Workstation (w/ WMware Tools)successfully? Thanks, Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- http://www.1and1.com/?k_id=8358073 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP to PTSN provider
Very true. I have found the outgoing CID to be very ... useful. Although occasionally inconsistent on the remote party's end, even though voipjet's CDR shows the CID string that I sent. -Andy On 4/17/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote: I have to agree that voipjet is a good service. If only they had did's it would be even better, but I like the fact that outgoing cid works well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Greenberg Sent: Saturday, April 16, 2005 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VOIP to PTSN provider 800 numbers are free to the caller because the recipient pays the charge. Voipjet has no way to get paid anything for carrying the calls, hence they are unwilling to use their resources to move calls with no revenue. Can you blame them? :) /edg --On Saturday, April 16, 2005 9:44 PM +0100 Chris Hills [EMAIL PROTECTED] wrote: Andy Hamilton wrote: I use voipjet and am quite pleased. Good enough rates and no noticeable quality issues. http://www.voipjet.com Plus, you can even test it before you buy. On their pricing page, they have:- There are some providers who can terminate some, but not all, 1800 numbers for free. (If they could terminate all 1800 numbers for free, then we'd use them!) I don't understand - I thought all 1800 numbers were free? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Illegal instruction (core dumped)
On April 17, 2005 05:55 am, Tom Fanning wrote: Illegal instruction (core dumped) Sounds like you have compiled asterisk for a processor that is greater than the processor you're running on. I.e. compiled and told it to use P4 instructions when you're on a P3, or maybe even told it to use MMX on a Via processor... Have just shoved PROC=i586 in the Makefile along with some commenting to see what happens. It's compiling right now. It is indeed on a Via Epia board. Cheers Tom Worked like a charm. Posting this here for future reference. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI vs. VOIP DID's, is it worth it?
On 4/17/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote: Hello All, I have been trying a did company for a few days. I find the service decent, but sound quality only moderate. Rather than spending 35 or so for monthly with did, I am considering an isdn bri at this location. How much more stable and reliable is bri or pri versus a voip did service? I like the concept of a bri more, but I do not get cid generation. Would anyone suggest bri over voip where available? I must say, I prefer higher voice quality. If anyone finds bri to be worth it (at about 54/month plus usage) please let me know what you think. I'm kind of asking the same questions myself right now. I think it depends a lot on what you are planning on using voip for. I also think that you are going to see reliability go up and up over the next year or two, so you have to take that into account also as you plan your infrastructure. I think new installations should at least be voip capable. Right now I would not rely on voip 100% for something business critical. Personally I'm looking at using voip but having adequate pstn access as a backup, with the incoming DID numbers being able to automatically route to the pstn in case of failure.I know I can do this if my numbers are 800 numbers, but I've still not found a way to do this with local number DID's, although I'm still looking. Reliability on incoming lines is a lot more difficult to deal with then outgoing. As long as you * server has connectivity, you could have 4-5 different providers in your dialplan and have it cascade down through them on failure. Wish it was that easy with DID's. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware
Roman Volf wrote: Have you tried putting both access points on the same channel? Good suggestion. It now seems to roam between access points nicely, even while a call is in progress. Also, I found firmware v1.5.3 if anyone needs it, along with manuals that are quite a bit more in depth than the ones I had before. If you need the firmware or manauls, feel free to e-mail me off-list. -j ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E M signalling with WCTE11XP - not all calls go through
Title: a simple question . I have successfully installed and configured a WCTE11XP card to connect with a voice switch (Cisco VCO/4K). Also, I have the SIP connection working as well, where a call from the voice switch properly transfers to the SIP phone. The voice switch is normally set up for EM and I have the WCTE11XP card setup for EM (wink in the zaptel.conf; em in zapata.conf). The cards communicate fine with no carrier failure or alarms on either side and the calls go through. The problem is that I run into sporadic issues where the call does not complete to the SIP phone in some instances. The voice switch shows a recorded message - presumably an * message that the extension is not available. Does anyone know where to see detailed logs for both the ZAP card and *. The /var/log/asterisk/messages file is a good source for the * side - but is there anything else. I would love to enable logging for each of the channels on the ZAP card - that is my goal. The zt tools (ztmonitor, zttools, etc.) do not seem to give enough info. Any help is appreciated. If this is the wrong listing, please advise. Thank you. No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.15 - Release Date: 4/16/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unbelievable...
Sure sounds like a veiled threat to me. Post something they don't like and find your support ticket ignored or possibly your account closed? Oh well guess I won't be getting any support from livevoip anytime soon:) Straight from the network status page on their website... If you are working a trouble ticket with LiveVoip support and start posting to mailing lists or newsgroups you are just wasting your time. LiveVoip LLC will not respond to such postings which in many cases are done to push support teams. If anything it will slow your ticket or cause the case to be closed. Our techs work hard for you! They are not going to take abuse in any form. Posting to these lists is done by some as a way of trying to obtain faster support or vent frustrations. LiveVoip has a Zero interest in these actions and will respond per our Terms Conditions if required. Let our people help you. That is what they get paid for. Are they busy? Of course. Do they work long hours? Duh. Treat them nice and Say Thanks. You will get further by being part of solutions, not part of the problems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unbelievable...
snacktime wrote: Sure sounds like a veiled threat to me. Veiled? Looks pretty overt to me. Why do these folks always think they can treat their customers like , when this is a market that really does have competition? They're not the incumbents, for God's sake, who get to do whatever they want. Thanks for the useful warning. . . B. Post something they don't like and find your support ticket ignored or possibly your account closed? Oh well guess I won't be getting any support from livevoip anytime soon:) Straight from the network status page on their website... If you are working a trouble ticket with LiveVoip support and start posting to mailing lists or newsgroups you are just wasting your time. LiveVoip LLC will not respond to such postings which in many cases are done to push support teams. If anything it will slow your ticket or cause the case to be closed. Our techs work hard for you! They are not going to take abuse in any form. Posting to these lists is done by some as a way of trying to obtain faster support or vent frustrations. LiveVoip has a Zero interest in these actions and will respond per our Terms Conditions if required. Let our people help you. That is what they get paid for. Are they busy? Of course. Do they work long hours? Duh. Treat them nice and Say Thanks. You will get further by being part of solutions, not part of the problems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Park a call then hunt for a *willing* person
On April 17, 2005 11:36 am, Philip Warner wrote: FWIW, I finally got it going using queues. The queue has one member which is a Local/[EMAIL PROTECTED] number. The context it points to uses Dial with the M() option, and it all seems to work...MoH runs all the time, and the caller can leave the queue via voicemail. Would you mind posting your config (dialplan and macro)? This sounds interesting. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iaxcomm
On Thu, 14 Apr 2005 09:33:21 +0500, amna saleem [EMAIL PROTECTED] wrote: No actually i have successfully installed (from scratch) and been using asterisk for more than 4 months now...i have been using diax phone ...but i came across this iaxcomm just thought about transfering a calljust playing around ..but i can`t really get it working ... maybe i am not getting the one hint can u help thanx Prior to 1.0rc3, you had to hit the OK button in the dialog box to complete the transfer. Enter key did not work. Now anything but Cancel should work. On 4/13/05, amna saleem [EMAIL PROTECTED] wrote: Hi! I was using iaxcomm but due to some reason am not able to transfer calls to some other extensionwhat maybe the problem do i have to make some changes to my extensions.conf??or iax.conf to be able to transfer calls Thanks Amna ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unbelievable...
Unbelieavable, and utterly disgraceful. Anyone found responsible for establishing such a policy would quickly find their ass on the street in any organization that understands the first thing about customer service. One doesn't build or protect a business by threatening and bullying one's customers. If one is worried about the bad impression that complainers are giving about the operation, figure out WHY they are driven to such extremes and DO SOMETHING ABOUT IT. It isn't rocket surgery. The principles of running an effective customer service organization are well known and readily available to anyone. The mind boggles... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Sunday, April 17, 2005 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unbelievable... Sure sounds like a veiled threat to me. Post something they don't like and find your support ticket ignored or possibly your account closed? Oh well guess I won't be getting any support from livevoip anytime soon:) Straight from the network status page on their website... If you are working a trouble ticket with LiveVoip support and start posting to mailing lists or newsgroups you are just wasting your time. LiveVoip LLC will not respond to such postings which in many cases are done to push support teams. If anything it will slow your ticket or cause the case to be closed. Our techs work hard for you! They are not going to take abuse in any form. Posting to these lists is done by some as a way of trying to obtain faster support or vent frustrations. LiveVoip has a Zero interest in these actions and will respond per our Terms Conditions if required. Let our people help you. That is what they get paid for. Are they busy? Of course. Do they work long hours? Duh. Treat them nice and Say Thanks. You will get further by being part of solutions, not part of the problems. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.15 - Release Date: 04/16/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPP g729 x86_64
I'm curious, how are you licensing your codec? The source is open, but the codec usage licensing is not. I think you'll find that licensing it from Digium will be much simpler, not to mention their code will Just Work(tm) without any messing around. -d At 12:08 PM 4/17/2005, you wrote: Hi, I 'm using a server DL145 with AMD opteron processors, with TE410P Digium Quad-Span card. The server is running RHEL4 x86_64. And have problem to compile codec g729 from http://www.readytechnology.co.uk/open/g729/, but ipp sample speech code not problem compile with ia32 or em64t. use l_ipp_ia32_itanium_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/ia32_itanium/lib -lippscmerged -lippsrmerged -lippsmerged -lippcore -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Iand use from l_ipp_em64t_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/em64t/lib -lippscem64t -lippsrem64t -lippsem64t -lippcoreem64t -L/opt/intel/ipp41/em64t/sharedlib/linuxem64t -lguide -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Any thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729A codec amd64/intel x86-64 optimisation?
can someone tell me more about this? On Apr 14, 2005, at 17:55, Roy Sigurd Karlsbakk wrote: hi for what I can see on digium's site, there is an x86-64 optimised g.729a codec. is this particularly optimised for intel or amd? I wonder most about sse/3dnow/whatever, as AFAICR this is quite different between the two. roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extension dialing resistivity
Which file control extension dialing responsivity / timing? When someone dial my extension, and is not fast enough, asterisk announces that the extension is not valid (it happened to me too). I have a mixed of two and three digit extensions in dial plan. Which setting controls this behavior. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension dialing resistivity
Hi Joseph, Let me take a guess - the problem only occurs when dialling four digit extensions? I think you will find that your dial plan is matching the three digit extension and then dialling it straight away - Asterisk won't wait for a timeout before trying to follow the dial plan, as soon as it finds a match it will try and dial whatever you've told it to (whether an extension context exists or not). This means, for example, that if you dialed extension '1234' then Asterisk will try and dial '123' if it finds a matching pattern in the dial plan - even if the extension '123' is invalid. There are two ways around this - either re-configure your dial plan so Asterisk won't get confused between three digit and four digit extensions (starting them in different numbers is a good idea) or configure your SIP phones (assuming you are using SIP phones) not to use forward dialling (i.e. to dial after a pre-set delay. We usually do the latter, as most SIP phones allow you to use the hash key to tell the phone to 'hurry up and dial now'. If you want to get really funky you can also write your dial plan so that it waits for 'n' seconds between each digit, but who could be bothered? FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Joseph wrote: Which file control extension dialing responsivity / timing? When someone dial my extension, and is not fast enough, asterisk announces that the extension is not valid (it happened to me too). I have a mixed of two and three digit extensions in dial plan. Which setting controls this behavior. - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extension dialing resistivity
I think you are right. I'm using Sipura-3000 which is causing the problem; though I don't know which setting. I just double check my dial-plan and I don't have any sort of conflicts: 123 / 1234 where the first three digit would match any four digit in any dial plan. The problem only occurs when someone dial IN from PSTN line. When I dial very slow internally it works perfectly; but when I dial IN from PSTN line sometime it causes problem. -- #Joseph On Mon, 2005-04-18 at 11:59 +1200, Damian Funnell wrote: Hi Joseph, Let me take a guess - the problem only occurs when dialling four digit extensions? I think you will find that your dial plan is matching the three digit extension and then dialling it straight away - Asterisk won't wait for a timeout before trying to follow the dial plan, as soon as it finds a match it will try and dial whatever you've told it to (whether an extension context exists or not). This means, for example, that if you dialed extension '1234' then Asterisk will try and dial '123' if it finds a matching pattern in the dial plan - even if the extension '123' is invalid. There are two ways around this - either re-configure your dial plan so Asterisk won't get confused between three digit and four digit extensions (starting them in different numbers is a good idea) or configure your SIP phones (assuming you are using SIP phones) not to use forward dialling (i.e. to dial after a pre-set delay. We usually do the latter, as most SIP phones allow you to use the hash key to tell the phone to 'hurry up and dial now'. If you want to get really funky you can also write your dial plan so that it waits for 'n' seconds between each digit, but who could be bothered? FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Joseph wrote: Which file control extension dialing responsivity / timing? When someone dial my extension, and is not fast enough, asterisk announces that the extension is not valid (it happened to me too). I have a mixed of two and three digit extensions in dial plan. Which setting controls this behavior. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe application. I don't think it installed with the standard 'make install' command. If not, how do I accomplish this? Thanks, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IPP g723 and getting error when starting asterisk
The compilation of codec g723.1 was fine. After I have copied to /usr/lib/asterisk/modules and started the asterisk -c .. I get this below error before asterisk quit. Anybody had any idea on Intel codec 723.1 ? [codec_g723.so] = (G723.1/PCM16 (signed linear) Codec Translator, based on IPP) Illegal instruction [EMAIL PROTECTED] G723.1]# Ouch ... error while writing audio data: : Broken pipe Thanks ** C.M. Rahman Jr. IT Manager CCNP, MCSE SecuritySecure your self by securing your System CompTI Security Plus Certified CCS Internet http://www.ccsi.com 13706 Research Blvd. Suite 100 Austin, TX 78750 Tel: 512-257-2274 Ex: 115 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of denon Sent: Sunday, April 17, 2005 5:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IPP g729 x86_64 I'm curious, how are you licensing your codec? The source is open, but the codec usage licensing is not. I think you'll find that licensing it from Digium will be much simpler, not to mention their code will Just Work(tm) without any messing around. -d At 12:08 PM 4/17/2005, you wrote: Hi, I 'm using a server DL145 with AMD opteron processors, with TE410P Digium Quad-Span card. The server is running RHEL4 x86_64. And have problem to compile codec g729 from http://www.readytechnology.co.uk/open/g729/, but ipp sample speech code not problem compile with ia32 or em64t. use l_ipp_ia32_itanium_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/ia32_itanium/lib -lippscmerged -lippsrmerged -lippsmerged -lippcore -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Iand use from l_ipp_em64t_p_4_1_2 : gcc -shared -static -Xlinker -x -o bin/codec_g729.so samples/util_e.o samples/util_d.o samples/codec_g729.o api/decg729fp.o api/encg729fp.o api/owng729fp.o api/usc729fp.o -L/opt/intel/ipp41/em64t/lib -lippscem64t -lippsrem64t -lippsem64t -lippcoreem64t -L/opt/intel/ipp41/em64t/sharedlib/linuxem64t -lguide -lpthread -lm /usr/bin/ld: /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: relocation R_X86_64_32 against `__deregister_frame_info' can not be used when making a shared object; recompile with -fPIC /usr/lib/gcc/x86_64-redhat-linux/3.4.3/crtbeginT.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [bin/codec_g729.so] Error 1 Any thoughts? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
Jesse Guardiani wrote: On Sun, 17 Apr 2005 01:39:09 -0400, Karl J. Vesterling wrote: H.323 will not traverse NAT. Sorry... I know, I was a big proponent of it when H.323 was the only standard VoIP protocol out there. Probably because when it came out NAT wasn't even thought of. The problem is that the control channel in H.323 discloses the internal IP address, and the various connections attempt to connect to each other. So you wind up with problems like audio only in one direction, etc... I thought SIP had the same problem though. Can't this be solved with address translation inside asterisk? You know, like the externip, localnet, and nat=yes options in sip.conf? It should be possible with h323, if you have control over the NAT points, with linux iptables/netfilter and the h323 NAT and conntrack helper modules. (netfilter.org patch-o-matic) And you stated you don't have control over one of the NATs, so that'd be out anyway. (better to go a different direction anyway, IMHO) And yes, SIP has the same problem, although many clients (hard and softphones) and * can usually compensate for this. Or is it simply impossible due to limitations within the H.323 spec? It's difficult to find information about this sort of thing on the internet. H.323 is such a broad spec... Wait a sec... COME TO THINK OF IT! Why not run asterisk on your linux box that you are running GnomeMeeting on, and use it to convert between H.323 and IAX and SIP??? After all, it is a penguin... That's certainly a good alternative. I'm currently in the process of hacking up the latest linphone (1.0.1) to fix a few personal show-stoppers. If I can get it to the point that I like it, then I'll probably just go with linphone. But you're right. If it's took much work, then I'll probably just start running asterisk on my laptop to do H.323 to SIP conversions. Thanks for the suggestion! I hadn't thought of that yet. I'd been looking at things like the commercial sip323 program, but I hadn't thought of doing it with a local copy of asterisk. Wouldn't it be simpler (and less resources) to set up an openvpn tunnel between the client and the * box? (since you're talking about softphones - for hardphones obviously you'd need to tunnel from another box then NAT or bridge) With openvpn or another vpn/tunnel solution, you can either bridge the client and asterisk LANs, or just create 1-1 tunnels from the client machine (if it's a softphone) to the * box. Either way you don't need to worry about NATs. (I'm doing this now for one of our hardphones,with an openVPN tunnel between linux gateway routers at each end.) j disclaimer - I know linux routing and firewalling, but only have a few months exposure to VOIP... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Register two account at Broadvoice with one asterisk box
Hello all, I have asked this question of Broadvoice support and the following is their responce: John, Unfortunately we are not able to fully support asterisk. We refer customers to the Asterisk forums where users are quite well versed and some are affiliated with BroadVoice. The only thing that comes to mind is that you may have to specify different ports for each number. Thank you, BroadVoice Customer Care tried voip-inf.org and not getting responce (down? or just me?) I can call in to and out of * from either number/account that i have. The problem is i would like to answer with different prompts based on which account/number the called dialed but broadvoice sends the call as if it came from whichever account i register second. Executing Answer(SIP/xx1492-d5d8, ) in new stack This is the same regardless off the number i call. I have tried register = user:pass@sip.broadvoice.com:5060 for first line and user2:pass2@sip.broadvoice.com:5061 this does not work for me. Is it possible to register on different ports? relevant sip.conf [broadvoice] type=peer username=xx1405 fromuser=xx1405 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very [broadvoice2] type=peer username=xx1492 fromuser=xx1492 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice2 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very any help is much appreciated Thank you, John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?
I just want to make the simplest call in which an X-Lite calls another X-Lite via asterisk. Unfortunately I failed time and time again. If someone is kind enough to show me sample config files by which asterisk works well, it will help me a lot. Best regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Router Terrible Voice Quality
On Thu, 14 Apr 2005, Rod Bacon wrote: I have been frustrated by a variety of zyxel issues/products and have found the best solution for all of them lies in a cylindrical receptacle that sits beside my desk... I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a built in DSL modem, and a single FXS port. Decent little router, now that the latest firmware is out, but tcp and udp timeouts through NAT seem to be set a little low, so I lose SSH sessions. G729 and G711-Ulaw sound great through it, and it supports no-power pass-through to the analog line. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] S100I - competitive price?
On Wed, 13 Apr 2005, Kevin P. Fleming wrote: [ DELETED] Realistically, how cheaply can you put together a box with a T-1 card and a channel bank with 24FXS ports (even disregarding G.729 transcoding, which would add to the cost)? $700? $800? more? I can't say for sure, but if you wanted to use a decent speed machine, I'd expect that the PC+TE110P+channel bank solution would cost at least $900, and that's using a bargain-basement PC and a used channel bank. Or if you didn't need FXS ports, we could take an old PM-3 w/ 50 DSP's in it, and a pair of the Dallas Framers to build a T1 channel bank via Ethernet. ;) Seriously, the PM3 would make an awesome platform for an Ethernet to T1 or PRI channel bank. The core is an AMD 5x86 processor, it can take 16 megs of ram, it has the entire TDM architecture already built into it, and the old Modem cards have Lucent DSPs that could easily implement transcoding and G.168 echo cancellation. Best of all, the boxes are dirt cheap, and.. I know for a fact that the ComOS has been ported to GCC on Linux and can be built. I've talked with the engineer that wrote the drivers for Livingston, and he's been thinking of writing an IAX2 stack for it. Digium.. you listening? ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe
Matt Schwartz wrote: Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe application. I don't think it installed with the standard 'make install' command. If not, how do I accomplish this? MeetMe requires Zaptel. If you do not have Zaptel installed, MeetMe won't build. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Register two account at Broadvoice with one asterisk box
There are a couple ways to do this. Or shoiuld be anyway. One is by setting the context as you have done. The other is by setting the extension at the end of the register line and doing a goto(someplace,s,1) for one line and goto(someplaceelse,s,1) for the other all from the same context. If one doesnt work for any reason you may try the other to see if that does, if neither works then there is something wrong with your broadvoice setup somehow. On Mon, 2005-04-18 at 01:11 +, John Millican wrote: Hello all, I have asked this question of Broadvoice support and the following is their responce: John, Unfortunately we are not able to fully support asterisk. We refer customers to the Asterisk forums where users are quite well versed and some are affiliated with BroadVoice. The only thing that comes to mind is that you may have to specify different ports for each number. Thank you, BroadVoice Customer Care tried voip-inf.org and not getting responce (down? or just me?) I can call in to and out of * from either number/account that i have. The problem is i would like to answer with different prompts based on which account/number the called dialed but broadvoice sends the call as if it came from whichever account i register second. Executing Answer(SIP/xx1492-d5d8, ) in new stack This is the same regardless off the number i call. I have tried register = user:pass@sip.broadvoice.com:5060 for first line and user2:pass2@sip.broadvoice.com:5061 this does not work for me. Is it possible to register on different ports? relevant sip.conf [broadvoice] type=peer username=xx1405 fromuser=xx1405 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very [broadvoice2] type=peer username=xx1492 fromuser=xx1492 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice2 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very any help is much appreciated Thank you, John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZyXEL Router Terrible Voice Quality
On Sun, 17 Apr 2005, Greg Boehnlein wrote: On Thu, 14 Apr 2005, Rod Bacon wrote: I have been frustrated by a variety of zyxel issues/products and have found the best solution for all of them lies in a cylindrical receptacle that sits beside my desk... I've had pretty good luck with the Zoom X5V Voice Modem so far. It has a built in DSL modem, and a single FXS port. Decent little router, now that the latest firmware is out, but tcp and udp timeouts through NAT seem to be set a little low, so I lose SSH sessions. I bought a dozen and have had bad luck with them. I couldn't keep an ssh session for more than 15 seconds. Trying to update firmware turned two of them into paperweights. I couldn't get the FXS to ever register. Other than that, it looked like a good idea. Is the NAT timeout configurable now? -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cannot dial two phones using zap
So the Panasonic extension dialed by Zap/3/206 command will ring and Zap/4/221 will not ring at all, even before ext 206 is picked up? Yes, exactly. Zap/4/221 won't ring at all. If you have two extensions numbered 211 212, why are you using 206 and 221 in your Dial command? 211 212 is plugged to asterisk, for dialing purpose. 206 221 is the extension I want to dial to. I would try this: 1. Make sure either extension will ring all by itself. Yes, they do ring all by itself. 2. Ring both at the same time, but put them in the other order in the Dial() command and see if that makes a difference. I've tried this: exten = 3,1,Dial(Zap/3/206,10) exten = 3,2,Wait(2) exten = 3,3,Dial(Zap/4/221,10) exten = 3,4,Hangup Zap/3/206 won't hangup / timeout. It just keep ringing and won't stop. :) 3. Rather than having: channel = 3,4 try channel = 3 channel = 4 just for fun. Tried this. No difference. 4. I don't know much about that Panasonic PBX, but are you sure calling two lines at the exact same time isn't messing it up? Not sure. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk + OH323 + NAT + gnomemeeting
On Sun, 17 Apr 2005 21:24:30 +0200, Bruno Hertz wrote: Jesse Guardiani [EMAIL PROTECTED] writes: Wait a sec... COME TO THINK OF IT! Why not run asterisk on your linux box that you are running GnomeMeeting on, and use it to convert between H.323 and IAX and SIP??? After all, it is a penguin... That's certainly a good alternative. I'm currently in the process of hacking up the latest linphone (1.0.1) to fix a few personal show-stoppers. If I can get it to the point that I like it, then I'll probably just go with linphone. But you're right. If it's took much work, then I'll probably just start running asterisk on my laptop to do H.323 to SIP conversions. Thanks for the suggestion! I hadn't thought of that yet. I'd been looking at things like the commercial sip323 program, but I hadn't thought of doing it with a local copy of asterisk. If your only reason to stick to H323 is Gnomemeeting you could try other softphones as well. Especially, the XLite beta for Linux looks promising, and some people like SJphone for Linux. I don't know about X-Lite, but sjphone seems only to support OSS. One of my requirements is ALSA support. Thus linphone and gnomemeeting. But, interestingly, gnomemeeting seems to be the only client capable of full duplex audio using ALSA+DMIX+DSNOOP+ASYM. Also, SIP support for Gnomemeeting is underway, but development is slow. I'm constantly pointing out to them how much interest there is, but things still seem to take their time ... Finally, on a recent discussion about the future design of GM on their list, I was surprised to learn that quite a few people really use it for direct PC to PC video calls over the internet. So somehow, after extensive NAT and router fiddling I guess, direct calls apparently work even with H323 (there is already support built into GM for external IP address discovery, as you know, so those remarks about transmission of bogus IP addresses on H323 level probably don't really apply in this case). Yeah. It supports STUN too, which seems to be the silver bullet for SIP. So I'm thinking the problem is more asterisk related. That's why I asked about gnugk. It seems to have more NAT translation support than asterisk, but my attempts at a working config haven't worked so far. -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Register two account at Broadvoice with one asterisk box
I can call in to and out of * from either number/account that i have. The problem is i would like to answer with different prompts based on which account/number the called dialed but broadvoice sends the call as if it came from whichever account i register second. Executing Answer(SIP/xx1492-d5d8, ) in new stack This is the same regardless off the number i call. I have tried register = user:pass@sip.broadvoice.com:5060 for first line and user2:pass2@sip.broadvoice.com:5061 this does not work for me. Is it possible to register on different ports? relevant sip.conf [broadvoice] type=peer username=xx1405 fromuser=xx1405 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very [broadvoice2] type=peer username=xx1492 fromuser=xx1492 secret=sniped pass host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=broadvoice2 dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=very any help is much appreciated Thank you, John M from Trixter and reposted at bottom( for ease of information flow) There are a couple ways to do this. Or shoiuld be anyway. One is by setting the context as you have done. The other is by setting the extension at the end of the register line and doing a goto(someplace,s,1) for one line and goto(someplaceelse,s,1) for the other all from the same context. snip Thank you but... this did not help. the problem is that the calls all come in as if from the same account, whichever registers second. called first number and got: -- Executing Answer(SIP/xx1492-b2c7, ) in new stack which should have been xx1405 called second number and got: -- Executing Answer(SIP/xx1492-2f6a, ) in new stack which is correct and is second in register statement Does anyone know if Broadvoice passes the equivalent of DNIS and is there a way to capture that in * from a VoIP call? John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe
MeetMe is straight forward. Follow the steps for ztdummy and there you go conferencing Check out www.voip-info.org for more info Cheers, ~Vamsi On 4/18/05, Matt Schwartz [EMAIL PROTECTED] wrote: Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe application. I don't think it installed with the standard 'make install' command. If not, how do I accomplish this? Thanks, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?
It would be easier if you could get send us your sip.conf entry and confiuration made in x-lite Also, please let us know where exactly the problem is. Is it while registering the x-lite or during the call and the exact error messages. Cheers, ~Vamsi On 4/18/05, Abraham WEI [EMAIL PROTECTED] wrote: I just want to make the simplest call in which an X-Lite calls another X-Lite via asterisk. Unfortunately I failed time and time again. If someone is kind enough to show me sample config files by which asterisk works well, it will help me a lot. Best regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can anyone send me sample config files for asterisk and X-Lite?
Vamsi Pottangi wrote: It would be easier if you could get send us your sip.conf entry and confiuration made in x-lite Also, please let us know where exactly the problem is. Is it while registering the x-lite or during the call and the exact error messages. Cheers, ~Vamsi On 4/18/05, Abraham WEI [EMAIL PROTECTED] wrote: I just want to make the simplest call in which an X-Lite calls another X-Lite via asterisk. Unfortunately I failed time and time again. If someone is kind enough to show me sample config files by which asterisk works well, it will help me a lot. Best regards, Abe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It might be easier if you started with [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium G.729 vs. IPP G.729
Hi, Did anyone compare G.729 implementations (from Digium and the one based on IPP) on features, stability, quality and reliabilty? It would be intresting to know how they fair against each other. I could be wrong, but in my testing I did notice a bit more hiss on Digiums codec thein IPPs. Anyone? This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe
Matt Schwartz wrote: Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe application. I don't think it installed with the standard 'make install' command. If not, how do I accomplish this? Thanks, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can be very sure that Meetme works with 1.0.7, as I just did it. Check out - http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x291.html for how to configure ztdummy if you are not using and Zaptel cards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (FIXED) extension dialing responsivity
Fixed! Another way of doing this is to give customer extra seconds between numbers: ... exten = s,4,Background(afterhours-menu) exten = s,5,DigitTimeout,5 ; give them 5 seconds between digits exten = s,6,ResponseTimeout,10 ; give them 90 seconds to make a choice ... -- #Joseph On Mon, 2005-04-18 at 11:59 +1200, Damian Funnell wrote: Hi Joseph, Let me take a guess - the problem only occurs when dialling four digit extensions? I think you will find that your dial plan is matching the three digit extension and then dialling it straight away - Asterisk won't wait for a timeout before trying to follow the dial plan, as soon as it finds a match it will try and dial whatever you've told it to (whether an extension context exists or not). This means, for example, that if you dialed extension '1234' then Asterisk will try and dial '123' if it finds a matching pattern in the dial plan - even if the extension '123' is invalid. There are two ways around this - either re-configure your dial plan so Asterisk won't get confused between three digit and four digit extensions (starting them in different numbers is a good idea) or configure your SIP phones (assuming you are using SIP phones) not to use forward dialling (i.e. to dial after a pre-set delay. We usually do the latter, as most SIP phones allow you to use the hash key to tell the phone to 'hurry up and dial now'. If you want to get really funky you can also write your dial plan so that it waits for 'n' seconds between each digit, but who could be bothered? FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz Joseph wrote: Which file control extension dialing responsivity / timing? When someone dial my extension, and is not fast enough, asterisk announces that the extension is not valid (it happened to me too). I have a mixed of two and three digit extensions in dial plan. Which setting controls this behavior. - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Register two account at Broadvoice with one asterisk box
Thank you but... this did not help. the problem is that the calls all come in as if from the same account, whichever registers second. called first number and got: -- Executing Answer(SIP/xx1492-b2c7, ) in new stack which should have been xx1405 now that you mention it I had the same problem with stanaphone. This may not be ideal but for grins change only the sip proxy that you use (lax, chi, dca and there is an older one they used to have dunno if its still up). See if you use different proxies if it works. If it does and that is the only problem I suggest that its a asterisk problem. I had forgotten that with stanaphone I had the same exact problem where 2 accounds to the same sip proxy would result in an inbound call coming into the 2nd account. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 881 8487 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hangs pc
Good day all I installed asterisk on a pc with redhat 9 and a 4port bri eachtime a call comes in,iax,sip,pstn it just hangs the pc Top shows 75% of the cpu goes to asterisk? Any Idea why? Please Help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unbelievable...
It's safe to assume that this particular company is pretty much functionally illiterate given the tone and tact of the rest of their comms. They won't be around long. On Apr 17, 2005, at 2:58 PM, Rusty Shackleford wrote: Unbelieavable, and utterly disgraceful. Anyone found responsible for establishing such a policy would quickly find their ass on the street in any organization that understands the first thing about customer service. One doesn't build or protect a business by threatening and bullying one's customers. If one is worried about the bad impression that complainers are giving about the operation, figure out WHY they are driven to such extremes and DO SOMETHING ABOUT IT. It isn't rocket surgery. The principles of running an effective customer service organization are well known and readily available to anyone. The mind boggles... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Sunday, April 17, 2005 2:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unbelievable... Sure sounds like a veiled threat to me. Post something they don't like and find your support ticket ignored or possibly your account closed? Oh well guess I won't be getting any support from livevoip anytime soon:) Straight from the network status page on their website... If you are working a trouble ticket with LiveVoip support and start posting to mailing lists or newsgroups you are just wasting your time. LiveVoip LLC will not respond to such postings which in many cases are done to push support teams. If anything it will slow your ticket or cause the case to be closed. Our techs work hard for you! They are not going to take abuse in any form. Posting to these lists is done by some as a way of trying to obtain faster support or vent frustrations. LiveVoip has a Zero interest in these actions and will respond per our Terms Conditions if required. Let our people help you. That is what they get paid for. Are they busy? Of course. Do they work long hours? Duh. Treat them nice and Say Thanks. You will get further by being part of solutions, not part of the problems. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.15 - Release Date: 04/16/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dynamic callrouting and billing?
Hi everyone, I am trying to figure out a plan for dynamic call forwarding between multiple asterisk servers. I would be dealing with around 30 different extension prefixes, each handled by a distinct asterisk server. Is there a sort of dynamic call routing feature to accomplish this, or I would have to statically describe each extension prefix in extensions.conf (not that it's too much to do any way, but it would be better done dynamically) ? Also, is anyone aware of a free centralized billing solution that I can take a look at so I could possibly start working on my own? Cheers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic Dialplan - Turn VM on/off?
G'day. I've been working with * for some time now, but mostly from a enterprise perspective. I've just setup my own box at home and want to enable some more home user type functionality. Does anyone have a trick to allow the dynamic modification of the dialplan by users? I want the ability to switch voicemail on/off (or at least alter the timeout). In essence, I want to simulate the act of manually turning an answering machine on when you leave home (for my wife). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
