Re: [Asterisk-Users] Siemens TC35 GSM gateway
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Fri, 30 Sep 2005, Andrew Smith wrote: I have a TC35 and am keen to see if anyone has both voice and sms working from Asterisk through this device? Google tells me that a few people have theorised about it, I can't find anyone claiming to be doing it. What would be the best way to put it into practice? Build a new channel for it? It's probably not that hard. I thought about it a couple of months ago. The device is pretty easy to use, ut uses AT-commands for everything you could want to do. It doesn't seem to be possible to get the audio out of it via the RS-232 port, so you'll have to connect it to a soundcard. The best approach is probably to take most of the code from chan_alsa, and just add the serial-communication and AT-commands needed to talk to the TC35. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDPWWxckvkFeO3ANARArViAJ0XS2PAEEoZS0Pm5Vb+MDi/5p/cCgCfXN89 LWzwfhrmH/LxxqjkMSQ708Y= =gp1K -END PGP SIGNATURE- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TxFax - RxFax on same machine hangs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 17 Aug 2005, Steve Underwood wrote: Roger Schreiter wrote: How can I enable asterisk to fax to itsself? Well, it won't be the normal operation, but when allowing clients to fax, it can happen by chance, that someone faxes to another user on the same machine without knowing it. Thanks for any hints! If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will not. The timing for these programs comes from the received data. No data, no work. I can confirm that this problem appears on a call through the PSTN. My setup is: TxFax - Asterisk - E1 - Asterisk (same box) -RxFax Asterisk version 1.0.9 and spandsp version 0.0.2pre18 on debian woody (3.0). I sent you an email about it with some debug information a week or so ago. If you need it again, or need some other info I'll be happy to provide it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDAui2ckvkFeO3ANARAuJPAKC00b+lEeHz+mOfb8J/zOF7+YAwggCeLFrG KGkJxLFGCeBY6foyDqC1xGM= =J6zk -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TxFax - RxFax on same machine hangs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 17 Aug 2005, Steve Underwood wrote: Bartek Kania wrote: If the call really dialed out through a PSTN port and back in it should work. It is was a pure internal connection between 2 processes it will not. The timing for these programs comes from the received data. No data, no work. I can confirm that this problem appears on a call through the PSTN. My setup is: TxFax - Asterisk - E1 - Asterisk (same box) -RxFax Asterisk version 1.0.9 and spandsp version 0.0.2pre18 on debian woody (3.0). I sent you an email about it with some debug information a week or so ago. If you need it again, or need some other info I'll be happy to provide it. Did you put txfax in caller mode? Yes I did. This is a snippet from 'show channel' for the two channels: Name: Zap/3-1 Type: Zap ... Frames in: 5249 Frames out: 265 Time to Hangup: 0 Elapsed Time: 0h1m45s ... Application: RxFAX Data: /tmp/1123753288.12.tif Stack: 1 Blocking in: ast_waitfor_nandfds and Name: Zap/28-1 Type: Zap ... Frames in: 3123 Frames out: 430 Time to Hangup: 0 Elapsed Time: 0h1m3s ... Application: TxFAX Data: /usr/local/asterisk/var/spool/asterisk/faxspool//ff-psbj1x.tif|caller|debug Stack: 0 Blocking in: ast_waitfor_nandfds The console seems to indicate that the faxes start to communicate using the slow modems, and then hang after switching to a fast modem. Log is attached. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFDA0I5ckvkFeO3ANARAjLJAJ0eXELd2WjzGOy59ptkFEE3kiUJcQCgxF9P 3WgYpTG5b1BfA3yOVk3w9wc= =lNab -END PGP SIGNATURE-Slow carrier up Slow carrier down Slow carrier up CSI: 40 35 38 33 20 30 30 30 36 2d 30 34 2d 36 34 2b 20 20 20 20 20 CSI without final frame tag Remote fax gave CSI as: +xx-xx- xxx DIS: 80 00 ce f4 80 80 81 80 80 80 18 DIS with final frame tag In state 10 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 20ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Start sending document Start tx document Changed from phase 2 to 4 DCS: 83 00 c6 80 80 80 00 HDLC underflow in state 3 Changed from phase 4 to 6 DCS with final frame tag In state 9 Coarse carrier frequency 1699.85 (66) Training error 0.506731 Training succeeded (constellation mismatch 0.703194) Changed from phase 6 to 3 Slow carrier up T4 timeout in state 4 Start rx document Start rx page - compression 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] skype channel
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (GNU/Linux) iD8DBQFCgLlVWYjaxM2wIe4RAuSKAJ9VNMIO2h838Y2yXAFDAQaJOjPa3gCfeokZ Ghsrpa8Gp3pHt5/bUinZKUA= =fUgt -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Zap channel calling back after hangup (due to polarity CID detection)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Mon, 28 Feb 2005, Anders F Eriksson wrote: Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried various combinations of hanguponpolarityswitch and answeronpolarityswitch (and without them). That issue is fixed in the CVS HEAD version of asterisk. There are a couple of workarounds possible with 1.0.6. Check the bugtracker for the bug where it was implemented for more information. (sorry, don't remember the bug-number and don't have time to look it up right now). You might also try the following patch: http://evil.gnarf.org/creativity/asterisk/20050110-answeronpolarity.diff which is what is in HEAD. But I don't know if it will work against asterisk stable. If you know C it should be no problem adapting it though. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (GNU/Linux) iD8DBQFCJDdgWYjaxM2wIe4RAln/AKDWh1FCK0n8yXrz0vxd/1bqpzgNkACfe6Ix ewEGA8Idz6uUIOFIXRSTZnQ= =OUHe -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Zap channel calling back after hangup (due to polarity CID detection)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 1 Mar 2005, Anders F Eriksson wrote: That issue is fixed in the CVS HEAD version of asterisk. There are a couple of workarounds possible with 1.0.6. Check the bugtracker for the bug where it was implemented for more information. (sorry, don't remember the bug-number and don't have time to look it up right now). I'm running CVS-HEAD-02/28/05-23:18:39 at the moment, and it still happens. I've seen the bugs in Mantis, but the answeronpolarity doesn't seem to make any difference ... Could you post a debug-log of when it happens? (enable debug and verbose in logger.conf) /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (GNU/Linux) iD8DBQFCJFwfWYjaxM2wIe4RAg+PAJ9gDLlQpTeWzLdG9Uk5+8JV6C9toACgkuf+ ucvqoNqP0G2Aojct8F4kXwk= =94Yp -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Zap channel calling back after hangup (dueto polarity CID detection)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 1 Mar 2005, Anders F Eriksson wrote: I'm running CVS-HEAD-02/28/05-23:18:39 at the moment, and it still happens. I've seen the bugs in Mantis, but the answeronpolarity doesn't seem to make any difference ... Could you post a debug-log of when it happens? (enable debug and verbose in logger.conf) This is the log: Ok, it seems the zaptel-driver is a little sensitive in your case: snip Mar 1 13:57:22 VERBOSE[15246]: == Starting post polarity CID detection on channel 4 Mar 1 13:57:22 VERBOSE[15256]: -- Starting simple switch on 'Zap/4-1' Mar 1 13:57:22 DEBUG[15256]: Receiving DTMF cid on channel Zap/4-1 Mar 1 13:57:23 DEBUG[15256]: Exception on 19, channel 4 Mar 1 13:57:23 DEBUG[15256]: Got event Ring/Answered(2) on channel 4 (index 0) Mar 1 13:57:23 DEBUG[15256]: Setting IDLE polarity due to ring. Old polarity was 1 The problem is the Got event Ring/Answered(2) line. Normally, a ring should not be detected and the DTMF-cid times out and no incoming call is registered. Make sure you load wctdm with the parameter 'opermode=SWEDEN', it might help. You might also try to increase the 'RING_DEBOUNCE' define in wctdm.c. If you load wctdm with debug=1 it prints some useful information to the kernel log for tuning the RING_DEBOUCE. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (GNU/Linux) iD8DBQFCJIX7WYjaxM2wIe4RAmh+AJ9IUZ81zd2vztjGnOXmtoSpQvN/sQCghu5/ /zBjljG1pe0pm+/zIhfTnVY= =A7M1 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Zap channel calling back after hangup (duetopolarity CID detection)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tue, 1 Mar 2005, Anders F Eriksson wrote: Make sure you load wctdm with the parameter 'opermode=SWEDEN', it might help. You might also try to increase the 'RING_DEBOUNCE' define in wctdm.c. If you load wctdm with debug=1 it prints some useful information to the kernel log for tuning the RING_DEBOUCE. opermode=SWEDEN did unfortunately not help. I increased the RING_DEBOUNCE and recompiled zaptel, and now it doesn't call back. However it seems it tries, but times out now. --- Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' Mar 1 17:48:11 WARNING[23918]: chan_zap.c:5351 ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch --- That is the expected behaviour. =) It is so because when YOU hang up, asterisk hangs up the channel and destroys it. A moment later (actually up to 90sec in sweden) the pstn disconnects the call and signals a polarity reversal. That reversal causes the above. I have a solution but I don't know enough about chan_zap to do it, and don't have the time to learn. The solution is basically to wait(with a timeout) for a reversal indicating that the PSTN has terminated the call before destroying the channel. But I don't really know where to put it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (GNU/Linux) iD8DBQFCJM51WYjaxM2wIe4RAsHOAKCFNOI60GRwJVXNCgLQU2mnEAB26QCfXqhf LmnJceTjwtA2rpF8grveBwI= =SL+0 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Enabling/disabling zaptel echo-can from dialplan.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Is it possible to enable/disable the zaptel echo-canceller from the dialplan? The reason I ask is that I want the echocanceller active on all calls except when someone is sending a fax. The simplest way would be to disable it on incoming calls to the fax numbers and leaving it on on all other calls. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFB5rz/WYjaxM2wIe4RAjGPAJ9q4XiUjqAZzFcNXN/+wh+OT9hABQCffWhm vBNmT7tnvuf/ELgrzfQLA6o= =/I2/ -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reverse Battery Disconnect Supervision in X100P or TDM400P FXO
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, 28 Jul 2004, Luis Vazquez wrote: Is posible to make the Digium FXO cards detect disconnect supervision by polarity reversal instead of battery drop?? It is possible, but probably not as simple as just detecting the reversal during an active call, since many telcos also signal when the remote end answers by reversing polarity and this might be confused for a hangup. The X100P will not be able to do it from what I understand, but the FXO-modules do detect it. There is no driver support yet. I have some preliminary patches for it (check bug number 9 on http://bugs.digium.com). I have also implemented this for myself in sweden, and it (mostly) works. Find me on IRC i #asterisk (I go by the nick eGnarF) or email me privately and I might be able to help you. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBCOdaWYjaxM2wIe4RAixtAKCZDdZoeXqHO3VsPDV06AwIy7hy2gCfd9e4 fUfK4IRenwO+p6r+p3rwX+A= =CFP+ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 problems transfering back and forth between pbxes
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi! I have two pbxes connected using IAX2 and trunk=yes. I tried the following today: From pbx1 (using a phone connected to a TDM400P) I call pbx2 and log in via DISA. Then I continue by calling back to pbx1. This works, and pbx2 seems to transfer the call just fine and hangs up the channel. If I then call pbx2 again (ie the call would be pbx1-pbx2-pbx1-pbx2) it does not work anymore. The effects of this are that pbx2 uses 100% cpu until the call disconnects (which happens in about 20sec or so). I am running CVS head as of today on both boxes. The relevant error-messages are: pbx1: Lots and lots of: Jul 8 20:12:35 WARNING[9225]: chan_iax2.c:4893 socket_read: Received trunked frame before first full voice frame pbx2: Jul 8 20:12:57 WARNING[9225]: chan_iax2.c:1409 attempt_transmit: Max retries exceeded to host x.x.x.x on [EMAIL PROTECTED]/16387 (type = 2, subclass = 2, ts=65, seqno=2) I tested this since I will probably be setting up a system using two pbxes where we will be doing a lot of transfers between pbxes. I.e. person1 on pbx1 answers and talks, transfers to person2 on pbx2. Person2 talks some and then transfers back to person1. Is this a known bug? An unknown bug? Or user(me) error? /B - -- A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA7ZbXWYjaxM2wIe4RAkTNAJ42p2LX5RgCzq4yceIeZOJmJ/QxVgCeIp/g xkocU0GxrFLl4Vm8F1ai8ys= =9jaJ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P FXO problems
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi! I live in Sweden and I am having problems getting asterisk to properly detect when a caller hangs up. And yes, I DO have disconnect-supervision on my line. Also asterisk sometimes misinterprets the disconnect-signal as another incoming call. This usually happens if I hang up first and then when the caller hangs up, asterisk treats it as a new call. I have tested this with asterisk from cvs HEAD as of today, and the stable 1.0 version. zaptel driver is cvs head. Also as of today. I have also tested with both callprogress = yes and no, but no luck. I plugged in my multimeter into the phoneline to see what actually happens here. This is how a call is signalled in sweden: 1) Line polarity reverses This is to mark the beginning of the CallerID 2) CallerID is sent using DTMF Don't know the system used, but I remember seeing a post about it in the archives. 2) Polarity reversed again Probably to mark the end of the CallerID 3) Ring signal is sent 4) Called party answers the call' 5) Polarity reverses I don't know why this is. Probably some indication that the call is in progress. 6) Parties talk (actually, I talked to myself =)) 7) callER hangs up 8) Polarity reverses This is the disconnect-notification. It seems that each data-phase is started by a polarity reversal, and ends when polarity reverses again. Asterisk seems to interpret step 1 as a new incoming call, and will answer immediately if I dont have a wait()-statement in my extensions.conf. This has the interresting sideffect that asterisk mistakes the callerID for an extension dialed by the caller. Asterisk doesn't recognize step 8 as a disconnect-notification, and if the phoneline was already on hook (according to asterisk) it will misinterpret this step as a new incoming call. I hope this is enough for someone to help me or give me some pointers on where to look. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAz1OkWYjaxM2wIe4RApVoAJ9S3BLHQsaUK2uf+iJf3kHyfItB+wCdEA8d NGGng0Oqm7X/XH27U13jEY8= =YI6d -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P hangup / ringing detection problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi!. I am having problems with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I connect an ordinary phone to it the led on the phone flashes once when someone hangs up on me. Also, if the person that calls me hangs up AFTER me, asterisk seems to interpret that as another incoming call. Also, sometimes I get these error-messages: WARNING[29711]: Ring/Off-hook in strange state 6 on channel 1 Unfortunately I haven't found anything that causes it. My zaptel.conf, and zapata.conf are attached to this email. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAxgXJWYjaxM2wIe4RAtw0AJ9Dxe4XQwr0urJPjVww/Tw4r26+kwCfT4qi 3yps6j8KadxLJnLxJajpnoM= =FrXq -END PGP SIGNATURE-# # Zaptel Configuration File # fxsks=1 fxoks=2-3 loadzone = no #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no defaultzone=no [channels] language = se context = external group = 1 echocancel = yes echocancelwhenbridged = yes echotraining = yes busydetect = yes callprogress = yes immediate = no usecallerid = yes ;callerid = asreceived signalling = fxs_ks channel = 1 context = internal group = 2 immediate = no signalling = fxo_ks channel = 2-3
[Asterisk-Users] Asterisk + E100P in Sweden
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am planning a small PBX for the company I work for. And since the price of the hardware and E1 installation is a big deal for us I want to make sure everything will work. So I wonder if anyone here has experience using the E100P to connect to a telco in Sweden? Tele2 in our case. I want to know if it will work without any problems, or if there are any issues I should be aware of. Sincerely, Bartek Kania - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAvw4uWYjaxM2wIe4RAuIRAKDHALqt/kHyHzeGuS07tn6iupYjfACfTXJF hh48uSyFYSFU94IV1iGUxko= =61UM -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PlayTones problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten = 123,1,Answer exten = 123,2,PlayTones(Busy) exten = 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: -- Executing Answer([EMAIL PROTECTED]/4, ) in new stack -- Executing Playtones([EMAIL PROTECTED]/4, Busy) in new stack -- Executing Hangup([EMAIL PROTECTED]/4, ) in new stack == Spawn extension (icepage, 123, 3) exited non-zero on '[EMAIL PROTECTED]/4' May 29 20:00:10 NOTICE[21526]: channel.c:1478 ast_set_write_format: Unable to find a path from UNKN to GSM -- Hungup '[EMAIL PROTECTED]/4' I don't have any FXS-hardware so I can't try it with a real phone. I am running the (stable) asterisk version from cvs from 3 days ago. Sincerely, Bartek - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAuNDJWYjaxM2wIe4RAtS1AKCwMqmaKILwzLg9ZnKx0+uDEw5drwCdFqqv vUBt3kLL7jVDsnWVrKYGr9w= =P8PB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PlayTones problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sat, 29 May 2004, Hermann Wecke wrote: On Sat, 29 May 2004, Bartek Kania wrote: I am having problems with the PlayTones application and VoIP softphones. But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. My extensions.conf always include a wait statement after playtones. Try it: exten = 123,1,Answer exten = 123,2,PlayTones(Busy) exten = 123,3,Wait(2); play busy tone for 2 seconds exten = 123,4,Hangup Thanks a bunch! I feel a little stupid for missing something like this though =) But I still get the following message on the console: May 29 21:01:58 NOTICE[22550]: channel.c:1478 ast_set_write_format: Unable to find a path from UNKN to GSM Can I just ignore it? /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFAuN6hWYjaxM2wIe4RAjqKAKCM37PIM+7Bb1OdOJIzOsWfocA0nACgncga dPgH1FX+jHVW8S67fM9jpVU= =tFbW -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
