[Asterisk-Users] Softfax/spandsp Makefile.patch rxfax/txfax
I followed the instructions at http://www.opencall.org/instructions.html and http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html I was able to compile spandsp (./configure ; make ; make install), manually patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the supplied patch does not fit the actual CVS apps/Makefile After make clean ; make install, I received this error: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init':../include/asterisk/lock.h:214: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:214: error: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:214: error: for each function it appears in. Any help? (yes, I copied app_rxfax.c app_txfax.c app_dtmftotext.c to apps dir also) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Softfax/spandsp Makefile.patch rxfax/txfax
Search the mailing lists, this has been answered a million times. Edit the Makefile and remove the entries for both app_rxfax.o and app_txfax.o and it will compile fine. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Saturday, June 19, 2004 10:14 PM To: Asterisk Mailling List Subject: [Asterisk-Users] Softfax/spandsp Makefile.patch rxfax/txfax I followed the instructions at http://www.opencall.org/instructions.html and http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html I was able to compile spandsp (./configure ; make ; make install), manually patched asterisk apps/Makefile (/usr/src/asterisk/apps), as the supplied patch does not fit the actual CVS apps/Makefile After make clean ; make install, I received this error: gcc -O2 -g -Iinclude -I../include -c -o app_rxfax.o app_rxfax.c In file included from app_rxfax.c:14: ../include/asterisk/lock.h: In function `ast_mutex_init':../include/asterisk/lock.h:214: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) ../include/asterisk/lock.h:214: error: (Each undeclared identifier is reported only once ../include/asterisk/lock.h:214: error: for each function it appears in. Any help? (yes, I copied app_rxfax.c app_txfax.c app_dtmftotext.c to apps dir also) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
Re: [Asterisk-Users] Maximum retries exceeded w/SIP
Brian: Thanks! I looked through the list and didn't see a correlation between what I was seeing and those parameters. Must have missed it. Thanks for your help. Andy - Original Message - From: Brian K. West To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 11:49 PM Subject: Re: [Asterisk-Users] Maximum retries exceeded w/SIP Usage of externip= and localnet= are what you are looking for. These all have been covered more than once in the mailing list... Remember GOOGLE IS YOUR FRIEND!! :P bkw - Original Message - From: Andrew Sackheim To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 9:29 PM Subject: [Asterisk-Users] Maximum retries exceeded w/SIP I struggled with this for several hours tonight.Turns out that if you have an * machine behind NAT, you must put the PUBLIC address in the bindaddr in sip.confIf you don't put it in, the Contact: header contains the NATted address and the sip phone can't get back to *.I don't know what happens if you mix and match sip phones on the local network -- it might not work unless the sipphone uses the public address as well.Hope this helps as I see this thread come up again and again...Andy---Steve, Sure, I could put all my machines on the public Internet, but that defeats the purpose of having a firewall in the first place. As an alternative, I could only place the * server on the outside, but I'd rather not give the script-kiddies another box to pound. Steve Totaro wrote: Can you disable your firewall? i am about to start this phase of asterisk an would like help from one newbie to another. otherwise this newbie will let you know how i did it. - Original Message - From: "Brad Waite" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 20, 2003 9:07 AM Subject: [Asterisk-Users] Maximum retries exceeded w/SIP First of all, I'd like to send a big "thank you" to all the folks who have helped me get this far. Now on to the next problem. Here's my current network setup: The Big I ---+--- FreeBSD FW --- * (10.0.0.253) PC (10.0.0.1) | +--- Laptop (public IP) natd is set up with the following rules: redirect_port udp 10.0.0.253:1-2 1-2 redirect_port udp 10.0.0.253:5060 5060 * is set up with the demo/sandbox config. I'm using XLite as my SIP client and have configured it on PC to work with *. I'm able to do everything I've tried so far. I should, though - I'm on the inside. However, when trying to make a call from the outside (via Laptop), something's breaking. I've set up the SIP proxy in XLite to be the external interface on the firewall, and am able to log into the proxy without difficulty. And while I can begin conversations, I can't keep them going for long. For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I get most of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a demonstration server located at Di" - at which point it gets cut off. The console spits out the following error: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 12384 (Response) Any ideas what could be going on? My first guess is the firewall, but I can't figure out why some of the packets would get through while others apparently are not. I'm at a loss. Brad Waite aka HankPoacher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting
Hi, I've compiled h323 from CVS 1.0 of Asterisk. I've copied h323.conf to /etc/asterisk but still I'm not able to call Asterisk from Netmeeting (I can call demo of Asterisk witl sip clients, but not Netmeeting as h323 client) with error: The person you called cannot accept Netmeeting calls. and no entry in Asterisk logs. I'd kindly ask for few answers: - is there any howto on Asterisk and Netmeeting or other h323 devices ? - do I have to enable enything else beside copying h323.conf to /etc/asterisk ? Regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk rxfax(): One page gets two pages
Hi, so far I had our PRI line connected via Digium TE410P and ZAP channel to asterisk which worked perfectly. I now have a second line coming in through a H.323 gateway and chan_oh323. rxfax() still works and receives faxes with G.711 alaw codec, but I always get one empty first page via H.323 - a one-page-fax becomes a two-page-fax with one empty page in front of it. It seems the empty page is only one scanline long. I am using asterisk CVS head of 2004-06-01 and spandsp-0.0.1k. A logfile from the console is attached below. May I kindly ask you to have a quick look at it? Many thanks, Jan Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 33 30 31 33 39 37 20 31 32 38 20 39 34 2b 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: +49 821 793103 DCS: 83 00 86 10 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 5ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1700.05 (66) Training error 2.676352 Training succeeded (constellation mismatch 4.690531) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.02 (70) Training error 3.675856 Training succeeded (constellation mismatch 5.689911) Fast carrier trained Fast carrier down Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 0 (got 1590, expected 1728). Page 1 of /usr/local/asterisk/var/spool/fax/0821793103-070245609-1086347692.4.tif: 1 rows received 0 total bad rows 0 max consecutive bad rows Changed from phase 5 to 3 Slow carrier up TSI: 43 33 30 31 33 39 37 20 31 32 38 20 39 34 2b 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: +49 821 793103 DCS: 83 00 86 10 DCS with final frame tag In state 5 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 5ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1656.38 (6) Fast carrier down Fast carrier up Coarse carrier frequency 1700.16 (70) Training error 4.311988 Training succeeded (constellation mismatch 5.752445) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1700.05 (66) Training error 14.160908 Training succeeded (constellation mismatch 10.055431) Fast carrier trained Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 1120 (got 0, expected 1728). Page 2 of /usr/local/asterisk/var/spool/fax/0821793103-070245609-1086347692.4.tif: 1121 rows received 0 total bad rows 0 max consecutive bad rows Rx page end detected Changed from phase 5 to 3 Slow carrier up Slow carrier down Slow carrier up EOP: 2f EOP with final frame tag In state 5 Changed from phase 3 to 4 MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up DCN: fb DCN with final frame tag In state 8 Disconnecting Changed from phase 3 to 7 Changed from phase 7 to 8 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting
Hello- Have you compiled and installed the proper versions of OpenH323 and PWLib? (Before you compiled the h323 code.) See the instructions in ~/asterisk/channels/h323/README.. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: Sunday, June 20, 2004 5:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting Hi, I've compiled h323 from CVS 1.0 of Asterisk. I've copied h323.conf to /etc/asterisk but still I'm not able to call Asterisk from Netmeeting (I can call demo of Asterisk witl sip clients, but not Netmeeting as h323 client) with error: The person you called cannot accept Netmeeting calls. and no entry in Asterisk logs. I'd kindly ask for few answers: - is there any howto on Asterisk and Netmeeting or other h323 devices ? - do I have to enable enything else beside copying h323.conf to /etc/asterisk ? Regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_oh323: busy not correctly signalled
Hi,
I have asterisk connected to PSTN via H.323 gateway via chan_oh323.
Incoming calls to SIP extensions work, but SIP message 486 busy here from a
busy extension isn't correctly forwarded to H.323.
As a result, a caller from the H.323 side calling a busy SIP extension gets some
rings and then an irritating timeout with H.323 message 'no user responding'
instead of 'user busy'.
Asterisk knows the user is busy and jumps to the prio+101 extension. The CDR
also logs the call as 'busy'.
With Zap ISDN channels the following works nicely:
exten = 12345,1,Dial(SIP/${EXTEN},45,r)
exten = 12345,102,SetVar(PRI_CAUSE=17)
exten = 12345,103,Hangup
Can we do something similar with chan_oh323?
Many thanks,
Jan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting
Hi, thanks for response. I've followed instructions and just tried ohphone and it works on simple call to Asterisk. But still cannot call from NetMeeting with same error. I still don't know how to setup h323 clients (NetMeeting, OhPhone) to Asterisk as extensions. I've tried several things but no success. It just says Gatekeeper cannot be found and similar. So I'm still kindly asking for more info on how: - to setup netmeeting to be able to call Asterisk - to setup NetMeeting, OhPhone or other h323 clients to register with Asterisk and be one of its extensions... - to setup h323 clients to place calls through Asterisk... Regards, Robert. - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 2:39 PM Subject: RE: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting Hello- Have you compiled and installed the proper versions of OpenH323 and PWLib? (Before you compiled the h323 code.) See the instructions in ~/asterisk/channels/h323/README.. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: Sunday, June 20, 2004 5:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting Hi, I've compiled h323 from CVS 1.0 of Asterisk. I've copied h323.conf to /etc/asterisk but still I'm not able to call Asterisk from Netmeeting (I can call demo of Asterisk witl sip clients, but not Netmeeting as h323 client) with error: The person you called cannot accept Netmeeting calls. and no entry in Asterisk logs. I'd kindly ask for few answers: - is there any howto on Asterisk and Netmeeting or other h323 devices ? - do I have to enable enything else beside copying h323.conf to /etc/asterisk ? Regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATA186 v3.1 SIP - Attended transfer: NO JOY
Hi, Response below. In the meantime: I would REALLY appreciate comments from an ATA186 SIP user who can tell me: - How to transfer a call without using #-transfer - Preferably more or less like how we are used to transferring in a classic pbx system Noteworthy: - Which Asterisk version (CVS/CVS-HEAD/...) - Which ATA186 firmware Thanks, Florian -Original Message- I have a similar issue with Sipura using compact headers, but not with regular headers. I am working on reproducing with the latest CVS. Maybe you are using compact SIP headers on your ATA186? http://bugs.digium.com/bug_view_page.php?bug_id=0001843 I have not found any setting on the ATA that can make such a difference in approach. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Wednesday, June 16, 2004 12:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA186 v3.1 SIP - Attended transfer: NO JOY Hi, I'm still hassling with the consultative/attended transfer stuff. Someone please help me identify this A lot has already been said about the ATA186. Some report it works fine, others say it doesn't. Lets get clarity on this. My scenario is reasonably simple (I think) Phone A: SIP/video1 Phone B: SIP/werkkamer Phone C: IAX2/provider Phone A calls phone B, they chat: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged Call SIP/video1-e2a0 SIP/video1-e2a0 (pbx1202 1 ) Up Dial SIP/swissSIP/snomSIP/werkkamerIAX2/florianSIP/video1 2 active channel(s) Phone B hits flash and gets a dialtone. Dials a number and connects to phone C: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data IAX2[172.28.8.8:4569]/7 ( s1 ) Up Bridged Call SIP/werkkamer-2507 SIP/werkkamer-2507 (pbx4307076 2 ) Up Dial IAX2/provider/4307076 SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged Call SIP/video1-e2a0 SIP/video1-e2a0 (pbx1202 1 ) Up Dial SIP/swissSIP/snomSIP/werkkamerIAX2/florianSIP/video1 4 active channel(s) Phone A now hears music on hold. Phone B and C can chat. Phone B now hits flash again. All phones end in a three-way conversation: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data IAX2[172.28.8.8:4569]/7 ( s1 ) Up Bridged Call SIP/werkkamer-2507 SIP/werkkamer-2507 (pbx4307076 2 ) Up Dial IAX2/provider/4307076 SIP/werkkamer-91f5 (from-werkkamer 1 ) Up Bridged Call SIP/video1-e2a0 SIP/video1-e2a0 (pbx1202 1 ) Up Dial SIP/swissSIP/snomSIP/werkkamerIAX2/florianSIP/video1 4 active channel(s) Now the misery starts: If Phone B wants to back out of the conversation, it seems phones C and A are also disconnected. I've tried doing this with SIP firmwares, 2.15, 2.16, 3.0 and 3.1 and CVS HEAD as of today. Other people have claimed success: http://lists.digium.com/pipermail/asterisk-users/2003-August/0 18388.html Is this: http://lists.digium.com/pipermail/asterisk-users/2003-August/0 18414.html also related ? By the way, canreinvite=no as suggested by Mark in one of the slightly related conversations on bugs.digium.com does not help... I would really _love_ to know why this is and to see it fixed somehow. A bounty would be in order. Can anyone comment on this ?? On a related note: If the consultation ends in a failure (user unavailable or unable to talk) the way to back out is hitting flash once if the remote hung up (ata doesn't give any tone at that time??) or twice if you got voicemail. The remote (phone A) briefly hears this, as the first flash opens a three-way conversation with phones A, B and the voicemail. The second one then disconnects the voicemail again. Not really elegant (albeit useable). Is there a better way ? Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration problem
Hi Folks, I'm having problem with GS registering in Asterisk. My setup is the following: [1755] type=friend incominglimit=10 qualify=no nat=yes insecure=no secret=X dtmfmode=rfc2833 username=1755 host=dynamic canreinvite=no defaultip=192.168.0.1 context=sip-incoming I have dozens of phones running the above configuration. All GS-BT101. The problem is that some of those phones, in the other side of the world, only register themselves during the boot and become unreachable after some minutes not re-registering themselves periodically what would be the right process. Registration time is 5 min. Firmware version 1.5.0.0 Asterisk version is 7.2 Anyone has any clue? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting
Hi- Yes, in the past I've had problems with both Ohphone and NetMeeting, and H323. Can't remember which is which, but with one program, the handshaking worked well but no audio, and with the other got audio ok (usually) but handshaking was flakey. Now using a Cisco 5300 to call into my asterisk box. Testing h.323 on CVS head (yesterday's version). Handshaking works, but no audio out from the asterisk box. Will re-test and quantify before posting to the bug list. Have you tried everything with the latest CVS (Head)? Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: Sunday, June 20, 2004 6:24 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting Hi, thanks for response. I've followed instructions and just tried ohphone and it works on simple call to Asterisk. But still cannot call from NetMeeting with same error. I still don't know how to setup h323 clients (NetMeeting, OhPhone) to Asterisk as extensions. I've tried several things but no success. It just says Gatekeeper cannot be found and similar. So I'm still kindly asking for more info on how: - to setup netmeeting to be able to call Asterisk - to setup NetMeeting, OhPhone or other h323 clients to register with Asterisk and be one of its extensions... - to setup h323 clients to place calls through Asterisk... Regards, Robert. - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 2:39 PM Subject: RE: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting Hello- Have you compiled and installed the proper versions of OpenH323 and PWLib? (Before you compiled the h323 code.) See the instructions in ~/asterisk/channels/h323/README.. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: Sunday, June 20, 2004 5:13 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting Hi, I've compiled h323 from CVS 1.0 of Asterisk. I've copied h323.conf to /etc/asterisk but still I'm not able to call Asterisk from Netmeeting (I can call demo of Asterisk witl sip clients, but not Netmeeting as h323 client) with error: The person you called cannot accept Netmeeting calls. and no entry in Asterisk logs. I'd kindly ask for few answers: - is there any howto on Asterisk and Netmeeting or other h323 devices ? - do I have to enable enything else beside copying h323.conf to /etc/asterisk ? Regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Date Time Stamp with Caller ID
Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks!
Re: [Asterisk-Users] Maximum retries exceeded on call
Eric C. Snowdeal III wrote: after registering the phones correctly and receiving a 200 o.k. message i can connect to other registered softphones and pstn endpoints [ via an voicepulse account ], but after making the initial connection, i can't hear any sound and i get disconnected after getting the following error message: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 41270 prompted by a recent email to the group [1] about setting the bindaddr, i took a closer look at the sip messages being sent back and forth and noticed that the contact header was incorrectly set to 127.0.0.1 in the 200 o.k. message [2]. once i set the bindaddr to the * machine's public ip address everything worked fine and and contact header i.p. address was set correctly. what's odd, at least to me, is that unlike the recent email about a similar issue [1], my * box is on a non-natted, public ip address so i would have thought that keeping the default bindaddr (0.0.0.0) would have worked, but obviously it didn't. not sure how to interpret the dirth of responses, perhaps this was frighteningly obvious to everyone else. [1] http://lists.digium.com/pipermail/asterisk-users/2004-June/051375.html [2] RECEIVE TIME: 7548279 RECEIVE my.public.asterisk.ip:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP my.wan.ip:5060;rport;branch=z9hG4bKFE76D502C2BF11D8A741000D93C22AF4;received=my.wan.ip;rport=1029 From: snowdeal sip:[EMAIL PROTECTED];tag=1666554831 To: snowdeal sip:[EMAIL PROTECTED];tag=as7f7ed33f Call-ID: [EMAIL PROTECTED] CSeq: 43970 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 1800 Contact: sip:[EMAIL PROTECTED];expires=1800 Date: Sun, 20 Jun 2004 13:44:34 GMT Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR in AGI
Is there any way to handle CDRswith AGI?
RE: [Asterisk-Users] Date Time Stamp with Caller ID
Kubat, Philip [EMAIL PROTECTED] wrote: (Article auto-converted from unnecessary HTML to nice plain text.) Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. My phones have a built-in clock. The Cisco 7960s are configured to take their time from a NTP server. I have a couple of portable DECT phones connected to a Sipura SPA-2000. The phones allow the time to be set from within the setup menu, and the Sipura uses our local NTP server. Check whether your phones have a clock. All of mine do in one way or another, so I always get the correct time associated with the Caller*ID notices. It's possible that the time/date is also encoded into the Caller*ID signal. I haven't had cause to look into that. It's possible that the DECT phones ignore the local time and use the time provided by the Sipura (if the Caller*ID signal does indeed supply this information). Again, I haven't had cause to look into that. Check whether your ATA device can be configured to use a NTP server. Also check whether your soft phone, and the phone connected to your ATA, has a clock you can set. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
I've changed the zaptel.conf to set both as internal, and it now seems to work, which is backwards to the config I thought it should have been, I would have thought that the Telewest PRI would have been 1 and the GDK 0? Can somebody confirm that this is the correct definition for timing, if it's a +ve number then it's external clocking with the lowest 1 being the highest priority. All spans are clocked relative to the external source and the external source selected is the lowest priority number that is currently being clocked? I'll experiment some more. -Original Message- From: Yifang Dai [mailto:[EMAIL PROTECTED] Sent: 19 June 2004 03:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Let's try again, missed a line in the last reply... On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote: On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote: LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco provider) --snip--- Any ideas on where to start? This is most likely to be a timing issue. You need to make sure your asterisk is get timing from your telco, and provide timing for you gdk pbx. /etc/asterisk/zaptel.conf is the place to look. -- Yifang Dai Senior System Administrator Yarde Metals Inc 45 Newell St, Southington, CT 06010 (Phone) 860-406-6107; (FAX) 860-406-4060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
Steve, your config description (timing) does sound odd. Could you re-post your revised config files? Thanks Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:18 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk I've changed the zaptel.conf to set both as internal, and it now seems to work, which is backwards to the config I thought it should have been, I would have thought that the Telewest PRI would have been 1 and the GDK 0? Can somebody confirm that this is the correct definition for timing, if it's a +ve number then it's external clocking with the lowest 1 being the highest priority. All spans are clocked relative to the external source and the external source selected is the lowest priority number that is currently being clocked? I'll experiment some more. -Original Message- From: Yifang Dai [mailto:[EMAIL PROTECTED] Sent: 19 June 2004 03:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Let's try again, missed a line in the last reply... On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote: On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote: LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco provider) --snip--- Any ideas on where to start? This is most likely to be a timing issue. You need to make sure your asterisk is get timing from your telco, and provide timing for you gdk pbx. /etc/asterisk/zaptel.conf is the place to look. -- Yifang Dai Senior System Administrator Yarde Metals Inc 45 Newell St, Southington, CT 06010 (Phone) 860-406-6107; (FAX) 860-406-4060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival and asterisk
I would also like to know how to insert a pause if possible. A comma is
seen as | not surprisingly.
I have no idea why no quotes.
- Original Message -
From: S. William Schulz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004 10:24 PM
Subject: Re: [Asterisk-Users] Festival and asterisk
On Sat, Jun 19, 2004 at 08:40:36AM -0600, Rich Adamson wrote:
extension.conf
exten = 555,1,Answer
exten = 555,2,Festival('good morning')
exten = 555,3,Wait(2)
exten = 555,4,Hangup
What's the problem I'm facing? Thanks in advance.
Remove the quote marks... should be like ...Festival(good morning)
What is the difference between an argument with quotes, and one without?
I ask because on one page of the wiki, it says not to use them, yet on
another [2] it highlights the use of quotes.
I have installed Festival today and it has been working without quotes,
but I wonder about whether it will parse (and change inflection) if
there are commas, question marks, and or exclamation points present and,
if so, if having them in the argument to Festival() will cause issues
without quotes.
[1] http://www.voip-info.org/wiki-Asterisk_Festival_installation
[2] http://www.voip-info.org/wiki-Asterisk+cmd+Festival
SWS
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
They look odd to me for sure, I'm certain (99.9%) that Telewest would not clock off of us, but as far as I can see, the current config (which allows the GDK to send and receive faxes) has no external clocking??? Here's the current config: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 Here's the original config which I took to mean that Telewest provided clock and span2 clocked off span1? span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 (Span1 goes to Telewest - our Telco provider, span2 goes to our current PBX, a GDK-186) Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 20 June 2004 16:34 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Steve, your config description (timing) does sound odd. Could you re-post your revised config files? Thanks Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:18 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk I've changed the zaptel.conf to set both as internal, and it now seems to work, which is backwards to the config I thought it should have been, I would have thought that the Telewest PRI would have been 1 and the GDK 0? Can somebody confirm that this is the correct definition for timing, if it's a +ve number then it's external clocking with the lowest 1 being the highest priority. All spans are clocked relative to the external source and the external source selected is the lowest priority number that is currently being clocked? I'll experiment some more. -Original Message- From: Yifang Dai [mailto:[EMAIL PROTECTED] Sent: 19 June 2004 03:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Let's try again, missed a line in the last reply... On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote: On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote: LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco provider) --snip--- Any ideas on where to start? This is most likely to be a timing issue. You need to make sure your asterisk is get timing from your telco, and provide timing for you gdk pbx. /etc/asterisk/zaptel.conf is the place to look. -- Yifang Dai Senior System Administrator Yarde Metals Inc 45 Newell St, Southington, CT 06010 (Phone) 860-406-6107; (FAX) 860-406-4060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 newbie: cannot get Asterisk to answer to netmeeting
Robert Rozman wrote: Hi, thanks for response. I've followed instructions and just tried ohphone and it works on simple call to Asterisk. But still cannot call from NetMeeting with same error. I still don't know how to setup h323 clients (NetMeeting, OhPhone) to Asterisk as extensions. I've tried several things but no success. It just says Gatekeeper cannot be found and similar. chan_h323 is not a Gatekeeper, today. chan_h323 is currently only a gateway, so don't try to register to it. So I'm still kindly asking for more info on how: - to setup netmeeting to be able to call Asterisk - to setup NetMeeting, OhPhone or other h323 clients to register with Asterisk and be one of its extensions... - to setup h323 clients to place calls through Asterisk... Dive into the wiki and help yourself first, then questions. http://www.voip-info.org/ Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR in AGI
shabanip wrote: Is there any way to handle CDRs with AGI? There is no need to. Install or create a CDR backend. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call waiting from PSTN
I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a beep I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? Tx, Bogdan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Date Time Stamp with Caller ID
On Sun, 20 Jun 2004 10:45:00 -0400, Kubat, Philip [EMAIL PROTECTED] wrote: Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. As Matteo said, the caller ID date should be generated by the FXS device. In this case, the ATA 188 should be generating it. Make sure that your ATA 188 has the correct time. You can have it fetch the correct time by specifying the NTPIP variable in the configuration. This variable points to an NTP time server. -Shaun ___ Thanks everyone that worked.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk
Hi Steve, How bizarre, your config doesn't look like it should work too well and certainly doesn't look like it should improve your fax problem! I assume that pri_cpe is set for span1 and pri_net for span2 ? Maybe, just maybe, Telewest reconfigured your PRI to look for clocking from your CPE but I've not encountered that configuration before. Try and leave the current config up for as long as you can before you return it to production mode and watch the CLI/logs to see if you get any sporadic clock slips (within a couple of hours I'd expect at least one episode of messages). One last thought, did you bounce the system after you made the changes to zaptel.conf or did you just reload * ? HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:48 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk They look odd to me for sure, I'm certain (99.9%) that Telewest would not clock off of us, but as far as I can see, the current config (which allows the GDK to send and receive faxes) has no external clocking??? Here's the current config: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 Here's the original config which I took to mean that Telewest provided clock and span2 clocked off span1? span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 (Span1 goes to Telewest - our Telco provider, span2 goes to our current PBX, a GDK-186) Steve -Original Message- From: Storer, Darren [mailto:[EMAIL PROTECTED] Sent: 20 June 2004 16:34 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Steve, your config description (timing) does sound odd. Could you re-post your revised config files? Thanks Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Hanselman Sent: 20 June 2004 16:18 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Problems with faxing via TE405P/Asterisk I've changed the zaptel.conf to set both as internal, and it now seems to work, which is backwards to the config I thought it should have been, I would have thought that the Telewest PRI would have been 1 and the GDK 0? Can somebody confirm that this is the correct definition for timing, if it's a +ve number then it's external clocking with the lowest 1 being the highest priority. All spans are clocked relative to the external source and the external source selected is the lowest priority number that is currently being clocked? I'll experiment some more. -Original Message- From: Yifang Dai [mailto:[EMAIL PROTECTED] Sent: 19 June 2004 03:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with faxing via TE405P/Asterisk Let's try again, missed a line in the last reply... On Fri, Jun 18, 2004 at 09:55:24PM -0400, Yifang Dai wrote: On Fri, Jun 18, 2004 at 09:27:52PM +0100, Steve Hanselman wrote: LG GDK-186 PBX --PRI--- TE405P/Asterisk ---PRI--- Telewest (Telco provider) --snip--- Any ideas on where to start? This is most likely to be a timing issue. You need to make sure your asterisk is get timing from your telco, and provide timing for you gdk pbx. /etc/asterisk/zaptel.conf is the place to look. -- Yifang Dai Senior System Administrator Yarde Metals Inc 45 Newell St, Southington, CT 06010 (Phone) 860-406-6107; (FAX) 860-406-4060 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal
[Asterisk-Users] One way audio
Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the offending line looks like this: [202] type=friend username=202 secret=voip-analog0 host=dynamic context=from-sip reinvite=no canreinvite=no disallow=all allow=ulaw nat=0 It works fine when calling between internally, or when the SPA-2000 is the calling source, but if a call comes in on a zap channel, the one way audio problem appears. -- Seth et lux in tenebris lucet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Date Time Stamp with Caller ID
On Sun, 20 Jun 2004, Kevin Walsh wrote: It's possible that the time/date is also encoded into the Caller*ID signal. I haven't had cause to look into that. It's possible that the DECT phones ignore the local time and use the time provided by the Sipura (if the Caller*ID signal does indeed supply this information). Again, I haven't had cause to look into that. Hi, Looks like BT Caller ID does include time and date information. have a look at http://www.sinet.bt.com/227v3p4.pdf Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Midifyed-Prepaid-Application
Hello. I have compile asterisk with modifyed prepaid application and populated the database to! I have fill the card, cardtype, cid, country, countrycode, reselers. I have make a cid=22 and I have add a user with username and callerid 22. But I allways get prepaid-no-aaa. Any one could help me how to authenticate? Note: I want to bill my local clients registred to my asterisk box using sip. Best Regards Hekuran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting from PSTN
On Jun 20, 2004, at 9:40 AM, [EMAIL PROTECTED] wrote: I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a beep I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? This is one of the catches I ran across while working on a new deployment of * in my home. The workaround I've been using is to hang up on the first call, at which point whatever was beeping on call waiting will ring the line. The drawback is that you have to hang up on the current call to get the call waiting call. On the 2.x Sipura firmware there's an option for the device to send the hook flash event as a SIP message, but * doesn't know what to do with it. Support for this could be added by, for example, triggering a hook flash on the zap channel currently in use by the phone that sent the message. I haven't tried this on other phones, or looked in to this in any depth. I'm also pretty new to *, so I don't know how hard it would be for me to make a patch that would handle this. That said, I could be totally wrong and there is a way to get the hook flash to translate from the phone to a zap channel, but as far as I can tell, there isn't. -- Seth cave quid dicis, quando, et cui Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Midifyed-Prepaid-Application
You have problems with pgsql. Check it. Regards, Gus - Original Message - From: Hekuran Doli [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 5:27 PM Subject: [Asterisk-Users] Midifyed-Prepaid-Application Hello. I have compile asterisk with modifyed prepaid application and populated the database to! I have fill the card, cardtype, cid, country, countrycode, reselers. I have make a cid=22 and I have add a user with username and callerid 22. But I allways get prepaid-no-aaa. Any one could help me how to authenticate? Note: I want to bill my local clients registred to my asterisk box using sip. Best Regards Hekuran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] enum problem with latest cvs
Hi, I posted an error message I was getting while using enum with the latest CVS, but the problem disappered. Well, it seems to be intermitten. The messages below: Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+16131234567$). Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 20 15:23:30 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error What is internesting is that this is happening only with 1 number, I have 2 other numbers registered and everything works fine with the other 2. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
Your regexp is WRONG 1.1.enum.blah.net naptr = 2 40 u iax2+E2U !^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1! . Thats a valid enum naptr record. It would translate into iax2:[EMAIL PROTECTED]/11 bkw - Original Message - From: Wojciech Tryc [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 2:27 PM Subject: [Asterisk-Users] enum problem with latest cvs Hi, I posted an error message I was getting while using enum with the latest CVS, but the problem disappered. Well, it seems to be intermitten. The messages below: Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+16131234567$). Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 20 15:23:30 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error What is internesting is that this is happening only with 1 number, I have 2 other numbers registered and everything works fine with the other 2. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival and asterisk
anyway now i can use the asterisk with the festival, it seems the problem is
the patch file festival-1.4.3.diff. in the patch file the festival directory
write down as festival-1.4.3 (included the version) but the actual festival
directory is festival (without any version info). so just rename the
festival folder to be festival-1.4.3 then apply the patch... Done...
(*)sorry for my bad english
Best Regards,
Freddy Setiawan
::Simple is Everything, Nothing is Complex::
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Totaro
Sent: Sunday, June 20, 2004 11:43 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Festival and asterisk
I would also like to know how to insert a pause if possible. A comma is
seen as | not surprisingly.
I have no idea why no quotes.
- Original Message -
From: S. William Schulz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004 10:24 PM
Subject: Re: [Asterisk-Users] Festival and asterisk
On Sat, Jun 19, 2004 at 08:40:36AM -0600, Rich Adamson wrote:
extension.conf
exten = 555,1,Answer
exten = 555,2,Festival('good morning')
exten = 555,3,Wait(2)
exten = 555,4,Hangup
What's the problem I'm facing? Thanks in advance.
Remove the quote marks... should be like ...Festival(good morning)
What is the difference between an argument with quotes, and one without?
I ask because on one page of the wiki, it says not to use them, yet on
another [2] it highlights the use of quotes.
I have installed Festival today and it has been working without quotes,
but I wonder about whether it will parse (and change inflection) if
there are commas, question marks, and or exclamation points present and,
if so, if having them in the argument to Festival() will cause issues
without quotes.
[1] http://www.voip-info.org/wiki-Asterisk_Festival_installation
[2] http://www.voip-info.org/wiki-Asterisk+cmd+Festival
SWS
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] enum problem with latest cvs
[EMAIL PROTECTED] wrote: Hi, I posted an error message I was getting while using enum with the latest CVS, but the problem disappered. Well, it seems to be intermitten. The messages below: Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+16131234567$). Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 20 15:23:30 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error What is internesting is that this is happening only with 1 number, I have 2 other numbers registered and everything works fine with the other 2. If that's the case, it's likely not a problem with Asterisk. Did you check the syntax of the regexp in the NAPTR record? Without knowing the number being looked up and the ENUM service being used, not much can be done to troubleshoot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Midifyed-Prepaid-Application
Try to debug postgresql query's done by app_prepaid... set log_statement = true in postgresql.conf and watch postgresql log's .. maybe it can help you ... Alex On Sun, 2004-06-20 at 20:27 +, Hekuran Doli wrote: bill my local clients registred to my asterisk box using sip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Midifyed-Prepaid-Application
Why not just use extension logic? (which is far more powerful than app_prepaid can ever dream of being) Jeremy McNamara Alexander Didebulidze wrote: Try to debug postgresql query's done by app_prepaid... set log_statement = true in postgresql.conf and watch postgresql log's .. maybe it can help you ... Alex On Sun, 2004-06-20 at 20:27 +, Hekuran Doli wrote: bill my local clients registred to my asterisk box using sip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
The number is 1-613-823-1716 and the enum service is e164.org. The most interesting part is that this is intermittent problem, sometimes it works sometimes it doesn't work. Again, any other lookups works just fine. Thanks, Wojtek - Original Message - From: Aaron J. Angel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 3:52 PM Subject: RE: [Asterisk-Users] enum problem with latest cvs [EMAIL PROTECTED] wrote: Hi, I posted an error message I was getting while using enum with the latest CVS, but the problem disappered. Well, it seems to be intermitten. The messages below: Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+16131234567$). Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 20 15:23:30 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error What is internesting is that this is happening only with 1 number, I have 2 other numbers registered and everything works fine with the other 2. If that's the case, it's likely not a problem with Asterisk. Did you check the syntax of the regexp in the NAPTR record? Without knowing the number being looked up and the ENUM service being used, not much can be done to troubleshoot. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
This is only intermittent problem!!?!? Wojtek - Original Message - From: Brian K. West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 3:50 PM Subject: Re: [Asterisk-Users] enum problem with latest cvs Your regexp is WRONG 1.1.enum.blah.net naptr = 2 40 u iax2+E2U !^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1! . Thats a valid enum naptr record. It would translate into iax2:[EMAIL PROTECTED]/11 bkw - Original Message - From: Wojciech Tryc [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 2:27 PM Subject: [Asterisk-Users] enum problem with latest cvs Hi, I posted an error message I was getting while using enum with the latest CVS, but the problem disappered. Well, it seems to be intermitten. The messages below: Jun 20 15:23:30 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+16131234567$). Jun 20 15:23:30 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 20 15:23:30 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 20 15:23:30 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error What is internesting is that this is happening only with 1 number, I have 2 other numbers registered and everything works fine with the other 2. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank Frustrations
I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime green. The T100P is sometimes red, sometimes green. The zaptel configuration is: span=1,0,0,esf,b8zs The channel bank settings are: Clock source: On or Off, seems not to make a difference Framing: ESF Line Code: B8ZS CSU: On or Off, seems not to make a difference Once I managed to get all three status lights on the channel bank green but the T100P was red. On the few cases the T100P gives a green status, zttool shows the RxB bits randomly flipping one and off. I have tried different T100P cards in different servers so that has been eliminated as a cause. I have made up several T1 loopback cables so I don't think it is a flaky cable. The remaining possibilities are: a) a bad channel bank (although it passes the self test) b) a bad zaptel configuration After several hours of trying different settings and DIP switches I am increasingly frustrated in trying to determine the cause of the problem. It has been especially difficult since power cycling the channel bank can result in a change to the status lights without any change to the settings or configuration. Any suggestions on what might be causing the problem or what to try next? We're trying to go live this week and this problem is critical. Thanks for any help/suggestions g. P.S. - What, exactly, is the meaning of the second argument to the zaptel span parameter (timing source) and how does it relate to the clock switch on the channel bank as I have tried all four combinations without obvious consistent affect? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream HT-286 // Custom Ring Tones
Hello, I am unsure if I was reading it wrong but someone once told me the ATA device Grandstream has supports custom ring tones, I have this device and have no idea how to implement this.. is it possible?? If so how do I do it? I am using the latest firmware.. Thanks! -- Stephen Rosebush, [EMAIL PROTECTED] http://www.desynched.org/ // PSTN USA:1-248-724-4452 x201 Netherlands:+31-(0)20-6598858 x63420 x201 // IP Phone FWD:63420 x201 IAXTEL: 1-700-356-6191 x201 SIP:sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk console mode
Hi
folks,
I use safe asterisk
to startup and run asterisk in the background. In Safe_asterisk script, there is
a parameter (right at the top ), CONSOLE which I can set to no or something. If
it is no asterisk startup as asterisk -vvvg , if it is set to something the
asterisk startup as asterisk -vvvg -c.
Now I am running an
agi script when calls get hung-up. That is in my
extensions.confIcall myagi.agilike h,1, agi,
myagi.agi. When I have asterisk started in console mode everything works
fine, however if I start -vvvg soon after the agi completes asterisk shut it
down.
== Spawn extension
(fwd-out-test, 613, 3) exited non-zero on 'SIP/-081467b0'
-- Executing AGI("SIP/-081467b0", "updatecb_post.agi") in new
stack -- Launched AGI Script
/var/lib/asterisk/agi-bin/updatecb_post.agi == Spawn extension
(fwd-out-test, h, 1) exited non-zero on
'SIP/-081467b0'asteriskremote*CLIDisconnected from Asterisk
serverExecuting last minute cleanupsAsterisk ending
(0).
Any Idea why this is
happening. ??? What are the pros and cons running asterisk in console mode
in safe asterisk ?
Cheers
DH
Re: [Asterisk-Users] Channel Bank Frustrations
George, We have this config working. Please give me a call (yeah, I'm at the office too) and we can walk through your config together. -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: George Pajari [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 5:51 PM Subject: [Asterisk-Users] Channel Bank Frustrations I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime green. The T100P is sometimes red, sometimes green. The zaptel configuration is: span=1,0,0,esf,b8zs The channel bank settings are: Clock source: On or Off, seems not to make a difference Framing: ESF Line Code: B8ZS CSU: On or Off, seems not to make a difference Once I managed to get all three status lights on the channel bank green but the T100P was red. On the few cases the T100P gives a green status, zttool shows the RxB bits randomly flipping one and off. I have tried different T100P cards in different servers so that has been eliminated as a cause. I have made up several T1 loopback cables so I don't think it is a flaky cable. The remaining possibilities are: a) a bad channel bank (although it passes the self test) b) a bad zaptel configuration After several hours of trying different settings and DIP switches I am increasingly frustrated in trying to determine the cause of the problem. It has been especially difficult since power cycling the channel bank can result in a change to the status lights without any change to the settings or configuration. Any suggestions on what might be causing the problem or what to try next? We're trying to go live this week and this problem is critical. Thanks for any help/suggestions g. P.S. - What, exactly, is the meaning of the second argument to the zaptel span parameter (timing source) and how does it relate to the clock switch on the channel bank as I have tried all four combinations without obvious consistent affect? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Midifyed-Prepaid-Application
any sample of extension logic around , just to have an idea? Best Regards Hekuran Why not just use extension logic? (which is far more powerful than app_prepaid can ever dream of being) Jeremy McNamara Alexander Didebulidze wrote: Try to debug postgresql query's done by app_prepaid... set log_statement = true in postgresql.conf and watch postgresql log's .. maybe it can help you ... Alex On Sun, 2004-06-20 at 20:27 +, Hekuran Doli wrote: bill my local clients registred to my asterisk box using sip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Midifyed-Prepaid-Application
Jeremy, Please explain little bit more how to use the extension logic for a prepaid app. Example would be greatly appreciated. Thanks --- Jeremy McNamara [EMAIL PROTECTED] wrote: Why not just use extension logic? (which is far more powerful than app_prepaid can ever dream of being) Jeremy McNamara Alexander Didebulidze wrote: Try to debug postgresql query's done by app_prepaid... set log_statement = true in postgresql.conf and watch postgresql log's .. maybe it can help you ... Alex On Sun, 2004-06-20 at 20:27 +, Hekuran Doli wrote: bill my local clients registred to my asterisk box using sip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No config file?
I updated from CVS yesterday and now everytime I start asterisk, I get the following message: config loader has no config file so nevermind. What does this mean? It doesn't seem to hurt anything, just a tad annoying to see everytime I run asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.
Ich werde ab 21.06.2004 nicht im Büro sein. Ich kehre zurück am 27.06.2004. Ich bin vom 20.6. bis 27.6 nicht per Email erreichbar und werde die Emails sobald als möglich ab dem 27.6 bearbeiten. Dringende Anfragen bitte an Andreas Widrig/CZWIAN/CH/Ascom machen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] enum problem with latest cvs
Brian K. West [EMAIL PROTECTED] wrote: Your regexp is WRONG The regexp isn't wrong. 1.1.enum.blah.net naptr = 2 40 u iax2+E2U !^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1! . 6.1.7.1.3.2.8.3.1.6.1.e164.org. 155 IN NAPTR 100 10 u E2U+IAX2 !^\\+16138231716$!iax2:[EMAIL PROTECTED]/16138231716! . So why would asterisk log an error with just !^+16138231716$? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No config file?
I'm having the same problem...nothing changed...just the CVS version. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sun, 20 Jun 2004, Aaron J. Angel wrote: I updated from CVS yesterday and now everytime I start asterisk, I get the following message: config loader has no config file so nevermind. What does this mean? It doesn't seem to hurt anything, just a tad annoying to see everytime I run asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question - TDM40B - Hunt Group Possibility??
Title: Message I was wondering if this is possible. I have a situation where I am connecting to a third party voicemail system from asterisk. I know this does not make since to everyone, but it has to be this way. Basically - I have an application that runs on the Asterisk system and when an employee calls into this system, they have an option to check there voicemail. This is where it needs to go over to the voicemail system. I would usually use an FXO card for this, but the other phone vendor I am working with is wondering is it possible to put the FXS cards I have in a hunt group - then I could call one of these ports and would ring the other voicemail system. If this can't be done that's fine - I have some FXO cards on order... Just thought I would check if anyone has ever done anything like this before. Thanks, Geoff Clark
Re: [Asterisk-Users] enum problem with latest cvs
E2U+IAX2 -- thats backwards also. bkw - Original Message - From: Aaron J. Angel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 6:13 PM Subject: RE: [Asterisk-Users] enum problem with latest cvs Brian K. West [EMAIL PROTECTED] wrote: Your regexp is WRONG The regexp isn't wrong. 1.1.enum.blah.net naptr = 2 40 u iax2+E2U !^\\+(.*)$!iax2:[EMAIL PROTECTED]/\\1! . 6.1.7.1.3.2.8.3.1.6.1.e164.org. 155 IN NAPTR 100 10 u E2U+IAX2 !^\\+16138231716$!iax2:[EMAIL PROTECTED]/16138231716! . So why would asterisk log an error with just !^+16138231716$? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need different contexts for 2 X100P FXO Cards and forwarding calls
Hi all, I have 2 incoming telephone lines connected to 2 X100P FXO Cards. One line is for my family, the other will be for my home office. I am new to Asterisk, but though that I would need calls being answered in different contexts. How can I direct one line to a given context and the other one to another, or is there a better way??? Thanks in advance, Francois ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
Brian K. West wrote: E2U+IAX2 -- thats backwards also. actually later RFC's specify it in that format... http://www.faqs.org/rfcs/rfc3762.html -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
Aaron J. Angel wrote: 6.1.7.1.3.2.8.3.1.6.1.e164.org. 155 IN NAPTR 100 10 u E2U+IAX2 !^\\+16138231716$!iax2:[EMAIL PROTECTED]/16138231716! . So why would asterisk log an error with just !^+16138231716$? I've probed all our name servers and they're all responding correctly, however someone else mentioned this the other day but we weren't able to track the problem then and assumed it was a bodgy resolver lib or something like that... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream CFG file generator
Adam Goryachev wrote: On Sat, 2004-06-19 at 06:13, Stephen R. Besch wrote: So, if someone could brief me on the GPL issue, and (perhaps someone else) offer a distribution point, it's free for the asking, VB sources and all. Stephen R. Besch Alright, I've waited a long time before offering this. Anyway, the short note is, if you want a central website for *anyone* to be able to upload a file to share with other users, etc etc.. specifically for Asterisk related (but perhaps off-topic for this list files) then visit http://www.websitemanagers.com.au/asterisk/ Uh, how about Sourceforge.net? That's what it's all about. It even helps you pick a license scheme. I'd highly suggest it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura config
This question isn't entirely Asterisk related, but I'm hoping that someone here may have the knowledge to respond to me anyway. I'm using Asterisk with several Sipura SPA-2000 SIP devices as FXS adapters. I would like to have my SPA's automatically provisioned through http or tftp, but I can't find any information on how to do so. Sipura's tech-support has not been very helpful in this matter, and I didn't purchase my Sipuras through an authorized retailer (once's from a VOIP provider, a couple from ebay, etc). Does anyone here have configuration information he/she could forward to me? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Data over Voice through Asterisk
Andrew Yager wrote: Hi, I'm trying to make a dialup internet connection through my asterisk PBX. When I bipass the Asterisk box, I can get 51600bps. When I run through the asterisk box, I'm limited to about 21600bps. Modem connections over 33.6K require that there be only a single analog-digital conversion in the path between the ISP's modem and your modem. With Asterisk in the middle, you are converting the signal back to digital, and then to analog again. You will not be able to go above 33.6K with Asterisk in between, unless you get digital phone service from your telco. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
No its backwards from what I have read... I got most of my info from many sources and E2U+SIP vs SIP+E2U but then again what do I know.. I have only been using enum for about a year now. bkw - Original Message - From: Duane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 8:32 PM Subject: Re: [Asterisk-Users] enum problem with latest cvs Brian K. West wrote: E2U+IAX2 -- thats backwards also. actually later RFC's specify it in that format... http://www.faqs.org/rfcs/rfc3762.html -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
The sip+E2U specifies a service to be contacted by SIP through the use of an E.164 to URI (E2U) translation. Thats in one of the documents that I have.. it depends on the direction of the translation. bkw - Original Message - From: Brian K. West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 9:33 PM Subject: Re: [Asterisk-Users] enum problem with latest cvs No its backwards from what I have read... I got most of my info from many sources and E2U+SIP vs SIP+E2U but then again what do I know.. I have only been using enum for about a year now. bkw - Original Message - From: Duane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 8:32 PM Subject: Re: [Asterisk-Users] enum problem with latest cvs Brian K. West wrote: E2U+IAX2 -- thats backwards also. actually later RFC's specify it in that format... http://www.faqs.org/rfcs/rfc3762.html -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
3762 is for h323 only: RFC 3762 - Telephone Number Mapping (ENUM) Service Registration for H.323 2916 is a bit more general: RFC 2916 - E.164 number and DNS Then we have: RFC 2915 - The Naming Authority Pointer (NAPTR) DNS Resource Record I was pointing out that E2U+IAX2 was backwards.. but then again asterisk doesn't really care about that... at this point. bkw - Original Message - From: Duane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 9:48 PM Subject: Re: [Asterisk-Users] enum problem with latest cvs Brian K. West wrote: but then again what do I know.. I have only been using enum for about a year now. RFC's change, if you want to stick to the standards you have to keep up with them... 3762 2916 -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] please mail me wave.cc and tts.scm
Can someone email me pached file festival/lib/tts.scm and festival/src/arch/festival/wave.cc (for festival version 1.4.3)? my mail is [EMAIL PROTECTED] Thanks in advance. Best Regards, Freddy Setiawan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] please mail me wave.cc and tts.scm
You have to do more than that.. patch and compile it manually. gentoo users: export USE=+asterisk emerge festival bkw - Original Message - From: Freddy Setiawan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 10:52 PM Subject: [Asterisk-Users] please mail me wave.cc and tts.scm Can someone email me pached file festival/lib/tts.scm and festival/src/arch/festival/wave.cc (for festival version 1.4.3)? my mail is [EMAIL PROTECTED] Thanks in advance. Best Regards, Freddy Setiawan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] enum problem with latest cvs
Try RFC3761. It specifies E2U+spec under section 2.4.2. It obsoletes RFC2916, and nothing has superseded it yet. Brian K. West [EMAIL PROTECTED] wrote: 3762 is for h323 only: RFC 3762 - Telephone Number Mapping (ENUM) Service Registration for H.323 2916 is a bit more general: RFC 2916 - E.164 number and DNS Then we have: RFC 2915 - The Naming Authority Pointer (NAPTR) DNS Resource Record I was pointing out that E2U+IAX2 was backwards.. but then again asterisk doesn't really care about that... at this point. bkw - Original Message - From: Duane [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 9:48 PM Subject: Re: [Asterisk-Users] enum problem with latest cvs Brian K. West wrote: but then again what do I know.. I have only been using enum for about a year now. RFC's change, if you want to stick to the standards you have to keep up with them... 3762 2916 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] please mail me wave.cc and tts.scm
i'm using RH9. Yes, after you send the file, i'll recompile the festival again. I think something wrong with the patch file its in the festival-1.4.3.diff it said: diff -u -r festival-1.4.3/lib/tts.scm festival-1.4.3-asterisk/lib/tts.scm --- festival-1.4.3/lib/tts.scm 2003-01-09 07:39:22.0 -0800 +++ festival-1.4.3-asterisk/lib/tts.scm 2003-08-14 12:07:00.000 -0700 i run the command patch -p1 /usr/src/asterisk/contrib/festival-1.4.3.diff it said successfully patched (tts.scm and wave.scm). but when i try the system, no voice was generated. i think the problem not in the asterisk, but the festival, because in the festival_server.log it said: server Mon Jun 21 12:19:30 2004 : Festival server started on port 1314 client(1) Mon Jun 21 12:22:01 2004 : accepted from localhost client(1) Mon Jun 21 12:22:01 2004 : disconnected it looks like the festival doesnt process the request from asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian K. West Sent: Monday, June 21, 2004 11:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] please mail me wave.cc and tts.scm You have to do more than that.. patch and compile it manually. gentoo users: export USE=+asterisk emerge festival bkw - Original Message - From: Freddy Setiawan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 20, 2004 10:52 PM Subject: [Asterisk-Users] please mail me wave.cc and tts.scm Can someone email me pached file festival/lib/tts.scm and festival/src/arch/festival/wave.cc (for festival version 1.4.3)? my mail is [EMAIL PROTECTED] Thanks in advance. Best Regards, Freddy Setiawan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
Brian K. West wrote: I was pointing out that E2U+IAX2 was backwards.. but then again asterisk doesn't really care about that... at this point. Considering the voip-info.org site has it as that, and that's before we go on to mention the fact IAX2 isn't in any rfc, and that it's ENUM implementation still leaves a few things out that I'm trying to get fixed, but person/professional and other things have taken priority of late... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problem with latest cvs
Aaron J. Angel wrote: Try RFC3761. It specifies E2U+spec under section 2.4.2. It obsoletes RFC2916, and nothing has superseded it yet. Damn always seem to get these out by 1 errors ;) -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the confrontation between the stream and the rock, the stream always wins; not through strength, but through persistence. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR in AGI
so my question changes to: - How to create a CDR backend? - I want to run my own scripts. shabanip shabanip wrote: Is there any way to handle CDRs with AGI? There is no need to. Install or create a CDR backend. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modified Prepaid database
Hi all, I've compiled and start the modified prepaid with postgresql. I would like to ask if anyone can give me a sample account to be populate in the database. I also want to confirm if this correct. I created a database prepaid (createdb prepaid) and from the asterisk-prepaid(current modified prepaid application) database folder, I execute (psql prepaid prepaid.sql, psql prepaid prepaid-data-country.sql, and psql prepaid prepaid-data-countryprefix.sql). thanks a lot. regards -- www.bembang.com
