Re: [asterisk-users] Newbie, Question on making a PSTN call..
I understand the desire to try, but you are trying too hard. Getting a soft modem to work with Asterisk is. like trying to push a string up a 10 foot pipe. At the least, buy an inexpensive FXO device from someone like Grandstream and use it via Ethernet to work with Asterisk. If you have greater ambitions, buy any appropriate piece of hardware and start with that. Otherwise, You are going to have a lot of string in that pipe, before you see any come out the top. You won't get help on this because no one really knows how to do it or if it will work at all. I am trying to help, by getting you to try a better way. Good luck. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shiva Kumar Sent: Tuesday, June 16, 2009 12:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie, Question on making a PSTN call.. Need help pls..Anyone? On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] feature keys no longer work after a call has been parked
Hey folks, I can park a call with #70 after enabling that feature in features.conf. However, once I retrieve the call from the parking lot, #70 cannot be used to park it again. Worse yet, none of the keys defined in the featuremap work anymore, include blindxfer or automon. Any ideas what may be the problem? -- martin | http://madduck.net/ | http://two.sentenc.es/ man sagt nicht 'nichts!', man sagt dafür 'jenseits' oder 'gott'. - friedrich nietzsche spamtraps: madduck.bo...@madduck.net digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to use # as feature key prefix
Hi folks, I was using the following featuremap: blindxfer = *1 disconnect = *9 atxfer = *2 parkcall = *7 automixmon = *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I tried to swap * fro # in the featuremap: blindxfer = #1 disconnect = #9 atxfer = #2 parkcall = #7 automixmon = #0 Unforunately, I cannot seem to make any of the features happen, neither during a normal call, nor during a conference. I've tried with multiple phones. What could be the problem? -- martin | http://madduck.net/ | http://two.sentenc.es/ and if the cloud bursts, thunder in your ear you shout and no one seems to hear and if the band you're in starts playing different tunes i'll see you on the dark side of the moon. -- pink floyd, 1972 spamtraps: madduck.bo...@madduck.net digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
On Mon, 15 Jun 2009, Shiva Kumar wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. Go out and buy specific hardware. OpenVox are really cheap these days. Well under £100 for a card with an FXO interface now. Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] feature keys no longer work after a call has been parked
martin f krafft wrote: Hey folks, I can park a call with #70 after enabling that feature in features.conf. However, once I retrieve the call from the parking lot, #70 cannot be used to park it again. Worse yet, none of the You fail to mention the version of Asterisk that you're working with. Under 1.4.20.1, I have a multi-parking patch that fixes this bug. I haven't had the need to upgrade to the newest version of the 1.4 series, so haven't looked to see if this patches is necessary. I can make the patch available on request. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] feature keys no longer work after a call has been parked
also sprach Doug Lytle supp...@drdos.info [2009.06.16.1142 +0200]: I can park a call with #70 after enabling that feature in features.conf. However, once I retrieve the call from the parking lot, #70 cannot be used to park it again. Worse yet, none of the You fail to mention the version of Asterisk that you're working with. Sorry. This is with the (experimental) Debian packages from Xorcom, version 1:1.6.1.0~dfsg-1.7248 I can make the patch available on request. Yes, please. It's good to know that this is a known bug. -- martin | http://madduck.net/ | http://two.sentenc.es/ (on the statement print 42 monkeys+1 snake) btw, both perl and python get this wrong. perl gives 43 and python gives 42 monkeys1 snake, when the answer is clearly 41 monkeys and 1 fat snake. -- jim fulton spamtraps: madduck.bo...@madduck.net digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Sun, Jun 14, 2009 at 03:10:03PM +1000, Alex Samad wrote: On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: [snip] The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware okay just for completeness in the thread, I have built the modules and it seems to be working, now to just re config asterisk -- I don't have people coming in the rope line saying, 'I'd like a new bridge, or how about some more highway money.' They're coming to say, 'I'm coming to tell you, Mr. President, I'm praying for you.' - George W. Bush 09/12/2006 Washington, DC said to journalists in the Oval Office (as reported by the National Review) signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Update Caller-ID after Dial()
Can you confirm that currently there is no way to update the caller ID via the manager interface once the B leg is ringing or connected? Looks like this would be feasible with the functions introduced in https://issues.asterisk.org/view.php?id=8824 ([patch] Remote (called) Party Identification - chan_sip chan_skinny implementation). Such functionality could be desirable in situations when a custom callerid number to name lookup takes more time than I am willing to spend before Dial()ing. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: [snip] Although I think I did see it download the firmware The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware Hi I have downloaded http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fw-vpmadt032-1.07.tar.gz and placed it in /usr/share/dadhi (un tarred ) when I reload the wctdm24xxp module I see [ 3350.588717] Found a Wildcard TDM: Wildcard TDM410P (4 modules) [ 3375.529165] Freed a Wildcard [ 3428.201978] Boosting ringer on slot 1 (89V peak) [ 3428.202059] Port 1: Installed -- AUTO FXS/DPO [ 3433.358455] Boosting ringer on slot 2 (89V peak) [ 3433.358474] Port 2: Installed -- AUTO FXS/DPO [ 3433.597478] Port 3: Not installed [ 3434.139532] Port 4: Installed -- AUTO FXO (AUSTRALIA mode) [ 3434.160522] VPM100: Not Present [ 3442.191181] Booting VPMADT032 [ 3446.075609] VPMADT032: Present and operational (Firmware version 117) but when I run dahdi_genconf it use the opensource echo canceller ! also can seem to find cahn_dadhi.so -- He was a state sponsor of terror. In other words, the government had declared, 'you are a state sponsor of terror.' - George W. Bush 01/23/2006 Manhattan, KS On Saddam Hussein signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] feature keys no longer work after a call has been parked
martin f krafft wrote:
Yes, please. It's good to know that this is a known bug.
Please remember, the patch is for 1.4
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
Index: res/res_features.c
===
--- res/res_features.c (revision 84404)
+++ res/res_features.c (working copy)
@@ -1670,7 +1670,7 @@
}
if (con) {
char
returnexten[AST_MAX_EXTENSION];
- snprintf(returnexten,
sizeof(returnexten), %s||t, peername);
+ snprintf(returnexten,
sizeof(returnexten), %s||tk, peername);
ast_add_extension2(con, 1,
peername, 1, NULL, NULL, Dial, strdup(returnexten), ast_free, registrar);
}
set_c_e_p(chan, parking_con_dial,
peername, 1);
@@ -1927,6 +1927,7 @@
memset(config, 0, sizeof(struct ast_bridge_config));
ast_set_flag((config.features_callee), AST_FEATURE_REDIRECT);
ast_set_flag((config.features_caller), AST_FEATURE_REDIRECT);
+ ast_set_flag((config.features_caller), AST_FEATURE_PARKCALL);
res = ast_bridge_call(chan, peer, config);
pbx_builtin_setvar_helper(chan, PARKEDCHANNEL, peer-name);
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[asterisk-users] Asterisk PSTN connection
Hi all, I am new to Asterisk and have a basic question. I am using Asterisk as PSTN gateway and want to connect Asterisk to a PSTN switch. We have ordered a Digium T1 card. I have installed the basic asterisk software and tried a call with dummy extension. Can someone please tell what all software i need to install to connect to PSTN switch along with installing digium T1 card ? Do I need any other hardware to do this? -Regards kavitha Date: Tue, 16 Jun 2009 21:04:37 +1000 From: a...@samad.com.au To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help building dahdi for debian On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: [snip] Although I think I did see it download the firmware The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware Hi I have downloaded http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fw-vpmadt032-1.07.tar.gz and placed it in /usr/share/dadhi (un tarred ) when I reload the wctdm24xxp module I see [ 3350.588717] Found a Wildcard TDM: Wildcard TDM410P (4 modules) [ 3375.529165] Freed a Wildcard [ 3428.201978] Boosting ringer on slot 1 (89V peak) [ 3428.202059] Port 1: Installed -- AUTO FXS/DPO [ 3433.358455] Boosting ringer on slot 2 (89V peak) [ 3433.358474] Port 2: Installed -- AUTO FXS/DPO [ 3433.597478] Port 3: Not installed [ 3434.139532] Port 4: Installed -- AUTO FXO (AUSTRALIA mode) [ 3434.160522] VPM100: Not Present [ 3442.191181] Booting VPMADT032 [ 3446.075609] VPMADT032: Present and operational (Firmware version 117) but when I run dahdi_genconf it use the opensource echo canceller ! also can seem to find cahn_dadhi.so -- He was a state sponsor of terror. In other words, the government had declared, 'you are a state sponsor of terror.' - George W. Bush 01/23/2006 Manhattan, KS On Saddam Hussein _ Missed any of the IPL matches ? Catch a recap of all the action on MSN Videos http://msnvideos.in/iplt20/msnvideoplayer.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x
Hi it seems like chan_dahdi.so is missing in debian asterisk 1.4.21 so I have upgraded to 1.6 and no I can load chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref [Jun 16 21:22:30] WARNING[4360]: loader.c:653 load_resource: Module 'chan_dahdi.so' could not be loaded. for a simple change over, its turning out to be not so simple -- A dictatorship would be a heck of a lot easier, there's no question about it. - George W. Bush 07/27/2001 signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote: On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: [snip] Although I think I did see it download the firmware The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware Hi I have downloaded http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fw-vpmadt032-1.07.tar.gz and placed it in /usr/share/dadhi (un tarred ) when I reload the wctdm24xxp module I see [ 3350.588717] Found a Wildcard TDM: Wildcard TDM410P (4 modules) [ 3375.529165] Freed a Wildcard [ 3428.201978] Boosting ringer on slot 1 (89V peak) [ 3428.202059] Port 1: Installed -- AUTO FXS/DPO [ 3433.358455] Boosting ringer on slot 2 (89V peak) [ 3433.358474] Port 2: Installed -- AUTO FXS/DPO [ 3433.597478] Port 3: Not installed [ 3434.139532] Port 4: Installed -- AUTO FXO (AUSTRALIA mode) [ 3434.160522] VPM100: Not Present [ 3442.191181] Booting VPMADT032 [ 3446.075609] VPMADT032: Present and operational (Firmware version 117) but when I run dahdi_genconf it use the opensource echo canceller ! dahdi_genconf generates configuration. It is a tool intended to help you and not a required step. It defaults to using mg2[1]. You can tell it to use a different echo canceller in the configuration it generates by setting 'echo_can hpec' in /etc/dahdi/genconf_parameters http://docs.tzafrir.org.il/dahdi-tools/#_sample_genconf_parameters [1] Actually, in the Debian package I patched it to default to use OSLEC, as it's available there and works better than mg2. The patch is trivial: http://patch-tracking.debian.net/patch/series/view/dahdi-tools/1:2.2.0~rc3-1/echocan_oslec also can seem to find cahn_dadhi.so That is part of asterisk . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and google talk
On Monday 15 June 2009 11:44:27 pm Antoine Patte wrote: Hello, I try to set up a gateway gtalk to sip. I test asterisk 1.6.1 and 1.4.21 from debian repository and the result is identical : no sound during the call. I had the same issue, only from gtalk to asterisk with some connections and not others.. Asterisk to gtalk works fine here. Have you allowed the RTP ports past your firewall? -- Thanks, Michael Maxwell eMail: metalm...@gmail.com Phone: +61 (03) 8680 4946 Web: mikey.webhop.org Powered By: PCBSD.org | FreeBSD.org | OpenSource.org PRAIL.org - Australia-Wide radio communications for free Sponsor: Hightek Hosting - A New Wave in IT and Hosting Technology Hosting, IT services, sales and onsite support! 1300 85 34 30 - www.hightekhosting.com.au ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x
Alex Samad wrote: it seems like chan_dahdi.so is missing in debian asterisk 1.4.21 so I have upgraded to 1.6 and no I can load chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref Moving from Asterisk 1.4 to 1.6 is not a 'simple changeover' :-) It's a major upgrade. In spite of that, for the time being chan_dahdi in Asterisk 1.6 requires that res_smdi be loaded first. If you are manually loading modules (instead of using 'autoload=yes' in /etc/asterisk/modules.conf'), you need to ensure that you load res_smdi before chan_dahdi. There are other modules that have dependencies like this as well, you can see what they are by running 'make menuselect' and looking at the dependency list for each module you want to load. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] feature keys no longer work after a call has been parked
also sprach Doug Lytle supp...@drdos.info [2009.06.16.1314 +0200]: Please remember, the patch is for 1.4 Right, and I found the corresponding lines in 1.6. But there are more questions now: - snprintf(returnexten, sizeof(returnexten), %s||t, peername); + snprintf(returnexten, sizeof(returnexten), %s||tk, peername); This suggests to me that transfers should work without your patch, but they don't. They work before parking, but after parking, transfers do not work. On the other hand, the use of lower-case tk suggests to me that this is for the callee and thus might be the flags in effect when the parking lot calls you back to remind you about a potentially forgotten call after 45 seconds. I went to try it and noticed two things: == Timeout for SIP/piper-01308f98 parked on 701 (default). Returning to park-dial,SIP0e71,1 -- Executing [sip0...@park-dial:1] Dial(SIP/piper-01308f98, SIP/e71|30|TKtk) in new stack [Jun 16 13:46:16] WARNING[13555]: pbx.c:953 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/e71|30|TKtk)) [Jun 16 13:46:37] WARNING[13555]: chan_sip.c:4526 create_addr: No such host: e71|30|TKtk [Jun 16 13:46:37] WARNING[13555]: app_dial.c:1518 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) First, 1.6 seems to include the patch or similar, since it includes the k flag. Second, the parking callback feature seems broken in 1.6. In the end, it seems that when I dial 701 to pick up the call, the dial flags of the original channel aren't restored. I don't know how to verify or further debug this though. Cheers, -- martin | http://madduck.net/ | http://two.sentenc.es/ literature always anticipates life. it does not copy it, but moulds it to its purpose. the nineteenth century, as we know it, is largely an invention of balzac. -- oscar wilde spamtraps: madduck.bo...@madduck.net digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x
On Tue, Jun 16, 2009 at 07:03:06AM -0500, Kevin P. Fleming wrote: Alex Samad wrote: it seems like chan_dahdi.so is missing in debian asterisk 1.4.21 so I have upgraded to 1.6 and no I can load chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref Moving from Asterisk 1.4 to 1.6 is not a 'simple changeover' :-) It's a major upgrade. In spite of that, for the time being chan_dahdi in Asterisk 1.6 requires that res_smdi be loaded first. If you are manually loading modules (instead of using 'autoload=yes' in /etc/asterisk/modules.conf'), you need to ensure that you load res_smdi before chan_dahdi. There are other modules that have dependencies like this as well, you can see what they are by running 'make menuselect' and looking at the dependency list for each module you want to load. ta, well I have banged my head against the wall for a bit and all seems to be working. for some reason i had no_load res_smdi which caused the problem. some question I have now is when i do a dahdi show channel 1 i get these interesting results Echo Cancellation: 128 taps currently OFF I have a hardware echo can and I have asked for it to be turned on ! Default law: ulaw I have a alaw:1-4 in the conf file, but it doesn't seem to take my last bug bear (maybe bug) is core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16 g723 - 56003 12001 1200128001 12001 12000 48002 164010 300018 - 28001 20001 40002 gsm 312020 - 16002 1600232002 16002 16001 52003 168011 304019 - 32002 24002 44003 ulaw 296020 44004 - 116002 2 1 36003 152011 288019 - 16002 8002 28003 alaw 296020 44004 1 -16002 2 1 36003 152011 288019 - 16002 8002 28003 g726aal2 312020 60004 16002 16002- 16002 16001 52003 168011 304019 - 1 24002 44003 adpcm 296020 44004 2 216002 - 1 36003 152011 288019 - 16002 8002 28003 slin 296019 44003 1 116001 1 - 36002 152010 288018 - 16001 8001 28002 lpc10 320022 68006 24004 2400440004 24004 24003 - 176013 312021 - 40004 32004 52005 g729 336021 84005 40003 4000356003 40003 40002 76004 - 328020 - 56003 48003 68004 speex 336022 84006 40004 4000456004 40004 40003 76005 192013 - - 56004 48004 68005 ilbc - - - -- - - - - - - - - - g726 312020 60004 16002 160021 16002 16001 52003 168011 304019 - - 24002 44003 g722 312020 60004 16002 1600232002 16002 16001 52003 168011 304019 - 32002 - 20001 slin16 332021 80005 36003 3600352003 36003 36002 72004 188012 324020 - 52003 20001 - the numbers seem to be way off, it seems to be able to do g729 - ulaw, my vsp only sends g729. but compared to 1.4 the numbers came back as single or maybe 2 digit - the g729 was 3 digits - I am on a soekris board, a amd geode machine. Something is I think going askew Thanks Alex -- I am a person who recognizes the fallacy of humans. - George W. Bush 09/19/2000 Oprah from Bush courts women in cozy 'Oprah' visit by William Goldshclag printed in the New York City edition of the Daily News, Sept. 20, 2000, page 5 signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Tue, Jun 16, 2009 at 02:35:08PM +0300, Tzafrir Cohen wrote: On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote: [snip] dahdi_genconf generates configuration. It is a tool intended to help you and not a required step. It defaults to using mg2[1]. You can tell it to use a different echo canceller in the configuration it generates by setting 'echo_can hpec' in /etc/dahdi/genconf_parameters http://docs.tzafrir.org.il/dahdi-tools/#_sample_genconf_parameters [1] Actually, in the Debian package I patched it to default to use OSLEC, as it's available there and works better than mg2. The patch is so how do I use the hardware echo can and how can I tell it is working ??? trivial: http://patch-tracking.debian.net/patch/series/view/dahdi-tools/1:2.2.0~rc3-1/echocan_oslec also can seem to find cahn_dadhi.so That is part of asterisk . not the 1.4.21 deb - but its in 1.6.1 -- That's just the nature of democracy. Sometimes pure politics enters into the rhetoric. - George W. Bush 08/08/2003 Crawford, TX signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Tue, Jun 16, 2009 at 02:35:08PM +0300, Tzafrir Cohen wrote: On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote: On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote: [snip] Although I think I did see it download the firmware The scripts for downloading the post-build firmware were moved to the separate dahdi-firmware package (sadly it has not made it into the archive yet). As the firmware files are not distributable I ended up just including the firmwares/ directory . That package is intended for non-free anyway (it includes the xpp firmwares) I included a script in that package to download and install the digium firmwares (a glorified make -C). /usr/share/dahdi/get-digium-firmware Hi I have downloaded http://downloads.digium.com/pub/telephony/firmware/releases/dahdi-fw-vpmadt032-1.07.tar.gz and placed it in /usr/share/dadhi (un tarred ) when I reload the wctdm24xxp module I see [ 3350.588717] Found a Wildcard TDM: Wildcard TDM410P (4 modules) [ 3375.529165] Freed a Wildcard [ 3428.201978] Boosting ringer on slot 1 (89V peak) [ 3428.202059] Port 1: Installed -- AUTO FXS/DPO [ 3433.358455] Boosting ringer on slot 2 (89V peak) [ 3433.358474] Port 2: Installed -- AUTO FXS/DPO [ 3433.597478] Port 3: Not installed [ 3434.139532] Port 4: Installed -- AUTO FXO (AUSTRALIA mode) [ 3434.160522] VPM100: Not Present [ 3442.191181] Booting VPMADT032 [ 3446.075609] VPMADT032: Present and operational (Firmware version 117) but when I run dahdi_genconf it use the opensource echo canceller ! dahdi_genconf generates configuration. It is a tool intended to help you and not a required step. It defaults to using mg2[1]. You can tell it to use a different echo canceller in the configuration it generates by setting 'echo_can hpec' in /etc/dahdi/genconf_parameters Duh. Ignore this. You asked about the hardware EC. The hardware EC can be activated regadrdless of the software EC you use. (Not sure exactly how. Anybody?) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x
Alex Samad wrote: some question I have now is when i do a dahdi show channel 1 i get these interesting results Echo Cancellation: 128 taps currently OFF I have a hardware echo can and I have asked for it to be turned on ! This is nothing new; the echo canceller on a channel is not enabled unless the channel is in an active call and the call hasn't caused it to be disabled (via tones). If you are looking at a channel that is inactive, this is exactly what you will see. Default law: ulaw I have a alaw:1-4 in the conf file, but it doesn't seem to take That is not valid syntax for /etc/dahdi/system.conf, correct syntax would be 'alaw=1-4'. my last bug bear (maybe bug) is core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) Note the phrase (in microseconds) here. This behavior has changed from Asterisk 1.4, and is documented in the documentation that came with Asterisk 1.6. If you haven't read the CHANGES and UPGRADE files thoroughly, you are spending time trying to understand things (assuming they are problems) that you don't need to spend. Something is I think going askew It sounds like you are trying to do a major upgrade without actually taking the time to learn what has changed and what that will require you to do. That's really very important, which is why we spend time writing that documentation in the first place :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no sdp or contact replacement using externip
Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. Regards, Ricardo. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no sdp or contact replacement using externip
Yes - we typically install behind NAT. The issue will usually be your firewall setup ...assuming you have setup your peers for NAT. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Martins Sent: Tuesday, June 16, 2009 9:40 AM To: Asterisk Users List Subject: [asterisk-users] no sdp or contact replacement using externip Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. Regards, Ricardo. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no sdp or contact replacement using externip
On Tue, 16 Jun 2009, Ricardo Martins wrote: Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. I have dozens of boxes doing it this way. All just work. Have you nat=yes in there too? Also you did port-forward from the router to the box as well, didn't you? Often the router will have a broke SIP ALG which will get in the way too. Turn it off if you can. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No exten available after pass between servers
Hi Rob, sip show users presents the following. Username Secret Accountcode Def.Context ACL NAT 2100 default No RFC3581 And that was what I needed! Thanks for the hint Rob, I just had to move the context around. On Tue, Jun 16, 2009 at 12:59 AM, Rob Hillisr...@hillis.dyndns.org wrote: Dan Pilcheck wrote: The call will go over the server fine, but when the Call Center server answer, the CLI returns: NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt from 10.0.10.20, request '2...@2xxx' does not exist What context are the phones in the extension range 2XXX in? I don't know what Vicidial's default context for extensions is, but I'd be surprised if it's 2XXX. Can you show us the results of sip show users after you remove the secret from the output? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.10: core restart on ReceiveFax()
sean darcy escribió:
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A fax machines is connected via an SPA 2102 on 173. Any calls from 173
are sent to:
[outbound-fax]
exten = 8447,1,Answer()
exten = 8447,n,GoSub(Capture-Fax,s,1)
exten =_NXXNXX,1,Answer()
exten =_NXXNXX,n,GoSub(DialOut-PSTN,s,1(1${EXTEN}))
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN}))
exten =_91NXXNXX,1,Answer()
exten =_91NXXNXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN:1}))
Actual outbound faxes work correctly. That is, a call from 173 to an
outside fax machine works.
The test faxes go to:
[Capture-Fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif) ;; 1.6 use ReceiveFAX
exten = s,n,Hangup()
When the test fax gets to ReceiveFax() asterisk restarts. Any calls at
the time are lost.
-- Executing [8...@outbound-fax:1] Answer(SIP/173-081d3780, )
in new stack
-- Executing [8...@outbound-fax:2] Gosub(SIP/173-081d3780,
Capture-Fax,s,1) in new stack
-- Executing [...@capture-fax:1] Set(SIP/173-081d3780,
FAXFILE=/var/spool/asterisk/fax/20090612_1710) in new stack
-- Executing [...@capture-fax:2] ReceiveFAX(SIP/173-081d3780,
/var/spool/asterisk/fax/20090612_1710.tif) in new stack
/var/spool/asterisk/fax exists, permissions 777:
ls -l /var/spool/asterisk
total 32
..
drwxrwxrwx 2 root root 4096 2009-05-03 14:21 fax
...
I've set debug and verbose to 20, but no further info.
What am I missing? Anybody have something like this working this working?
sean
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Two thoughts on this:
1. On Exten 8447, why are you using a GoSub instead of a simple Goto?
(Shouldn't pose a problem, just curious)
2. I wouldn't recommend the use of date-time only to form the filename
of the received fax. This was the key issue that was killing asterisk in
my case, when I switched to date-time + uniqueid for example, everything
went fine. That's because it looks like RxFax() or in this case
ReceiveFax() doesn't play well with duplicate (open) filenames, which I
don't know if they occur in your case. It definitely can happen if you
have programmed it and have the capacity to receive simultaneous faxes
at a time. After the change my machine is rock stable, receiving and
sending hundreds of faxes a day with no restarts.
Hope this helps.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] Help building dahdi for debian
Tzafrir Cohen wrote: Duh. Ignore this. You asked about the hardware EC. The hardware EC can be activated regadrdless of the software EC you use. (Not sure exactly how. Anybody?) It's automatic; nothing needs to be specified in /etc/dahdi/system.conf at all. If chan_dahdi is configured to request echo cancellation on the channel, and there is a hardware EC present (and not disabled via module parameters) it will be used. No software EC needs to be configured or even loaded. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to use # as feature key prefix
On Tue, Jun 16, 2009 at 9:56 AM, Danny Nicholasda...@debsinc.com wrote: The problem is the Asterisk Read function. It is set to accept as many 0-9 and * as you want to throw at it, then stop on # or timeout. Unless you disable the # stops, you can't use # in features. I would strongly caution against that because you would almost certainly break something else. Is that a configuration change or a let's to edit source code change? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to use # as feature key prefix
The problem is the Asterisk Read function. It is set to accept as many 0-9 and * as you want to throw at it, then stop on # or timeout. Unless you disable the # stops, you can't use # in features. I would strongly caution against that because you would almost certainly break something else. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of martin f krafft Sent: Tuesday, June 16, 2009 2:29 AM To: asterisk users mailing list Subject: [asterisk-users] Unable to use # as feature key prefix Hi folks, I was using the following featuremap: blindxfer = *1 disconnect = *9 atxfer = *2 parkcall = *7 automixmon = *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I tried to swap * fro # in the featuremap: blindxfer = #1 disconnect = #9 atxfer = #2 parkcall = #7 automixmon = #0 Unforunately, I cannot seem to make any of the features happen, neither during a normal call, nor during a conference. I've tried with multiple phones. What could be the problem? -- martin | http://madduck.net/ | http://two.sentenc.es/ and if the cloud bursts, thunder in your ear you shout and no one seems to hear and if the band you're in starts playing different tunes i'll see you on the dark side of the moon. -- pink floyd, 1972 spamtraps: madduck.bo...@madduck.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to use # as feature key prefix
LESC - proceed at your own peril... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Moore Sent: Tuesday, June 16, 2009 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unable to use # as feature key prefix On Tue, Jun 16, 2009 at 9:56 AM, Danny Nicholasda...@debsinc.com wrote: The problem is the Asterisk Read function. It is set to accept as many 0-9 and * as you want to throw at it, then stop on # or timeout. Unless you disable the # stops, you can't use # in features. I would strongly caution against that because you would almost certainly break something else. Is that a configuration change or a let's to edit source code change? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Resetting Marker Bits
Anyone have any idea on how to force marker bits on in RTP ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 10 June 2009 14:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Resetting Marker Bits (resend as apparently I was blocked) Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client - A*k1 - A*k2 - PSTN Provider/Gradwell - O2 - Mobile - the SIP client dials on O2 mobile, call goes out to A*1. - A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers and normal office phones. - A*k2 dials some local Cisco phones, then on no answer plays an audio file, so call is ANSWERED. - A*k2 then Dials out to gradwell, to a mobile phone number. - Gradwell takes the call, routes it via PSTN. My problem, is that at the point where the O2 mobile accepts the call, I get one-way audio. (SIP Client outbound, nothing inbound). Tracing the RTP stream all the way back, I can see that audio makes it all the way to the SIP Client. However, we notice that at the point where the O2 mobile answers, the TIME= value of the packet jumps significantly, say from 119248 to 1518324408. Talking to the sip client developer, they say that I need to enable SIP Markers on the server (I guess A*k2), so that if the stream source changes then the timers are reset. Does this sound right, and if so, how do I do that ? I am running an older load on A*K2 of 1.4.18, and 1.4.15svn (privately compiled to add an extra codec) on A*k1. I can look into upgrading these, but the developer thinks it's just a missing config on Asterisk. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update Caller-ID after Dial()
You can't change the actual callerid, but why not load a variable and update that? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Tuesday, June 16, 2009 6:02 AM To: Asterisk Users Subject: [asterisk-users] Update Caller-ID after Dial() Can you confirm that currently there is no way to update the caller ID via the manager interface once the B leg is ringing or connected? Looks like this would be feasible with the functions introduced in https://issues.asterisk.org/view.php?id=8824 ([patch] Remote (called) Party Identification - chan_sip chan_skinny implementation). Such functionality could be desirable in situations when a custom callerid number to name lookup takes more time than I am willing to spend before Dial()ing. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no sdp or contact replacement using externip
Yes Gordon. I'm using nat=yes and I don't have an ALG enabled router/firewall. I used the sip debug output on the asterisk(s) and could see the sdp headers as they were gererated by asterisk, with the wrong (internal) address on it. Asterisk is sending the audio to the correct way, the public IP of client side NAT. But the client is sending it to the wrong address, the private IP of asterisk side NAT. Rgrs, Ricardo. Gordon Henderson escreveu: On Tue, 16 Jun 2009, Ricardo Martins wrote: Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. I have dozens of boxes doing it this way. All "just work". Have you nat=yes in there too? Also you did port-forward from the router to the box as well, didn't you? Often the router will have a broke SIP ALG which will get in the way too. Turn it off if you can. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no sdp or contact replacement using externip
I'm not sure if your problem is addressed by this: https://issues.asterisk.org/view.php?id=14546 . If that's the case it was solved in version 1.4.25 Best regards, Santi 2009/6/16 Ricardo Martins rpopp...@gmail.com Yes Gordon. I'm using nat=yes and I don't have an ALG enabled router/firewall. I used the sip debug output on the asterisk(s) and could see the sdp headers as they were gererated by asterisk, with the wrong (internal) address on it. Asterisk is sending the audio to the correct way, the public IP of client side NAT. But the client is sending it to the wrong address, the private IP of asterisk side NAT. Rgrs, Ricardo. Gordon Henderson escreveu: On Tue, 16 Jun 2009, Ricardo Martins wrote: Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. I have dozens of boxes doing it this way. All just work. Have you nat=yes in there too? Also you did port-forward from the router to the box as well, didn't you? Often the router will have a broke SIP ALG which will get in the way too. Turn it off if you can. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update Caller-ID after Dial()
Philipp Kempgen wrote: Can you confirm that currently there is no way to update the caller ID via the manager interface once the B leg is ringing or connected? Looks like this would be feasible with the functions introduced in https://issues.asterisk.org/view.php?id=8824 ([patch] Remote (called) Party Identification - chan_sip chan_skinny implementation). Such functionality could be desirable in situations when a custom callerid number to name lookup takes more time than I am willing to spend before Dial()ing. Danny Nicholas schrieb: You can't change the actual callerid, but why not load a variable and update that? Setting a variable wouldn't send a caller-id update to the B leg. What am I missing? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update Caller-ID after Dial()
In the B leg, check for the variable value instead of Callerid(num). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Tuesday, June 16, 2009 10:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Update Caller-ID after Dial() Philipp Kempgen wrote: Can you confirm that currently there is no way to update the caller ID via the manager interface once the B leg is ringing or connected? Looks like this would be feasible with the functions introduced in https://issues.asterisk.org/view.php?id=8824 ([patch] Remote (called) Party Identification - chan_sip chan_skinny implementation). Such functionality could be desirable in situations when a custom callerid number to name lookup takes more time than I am willing to spend before Dial()ing. Danny Nicholas schrieb: You can't change the actual callerid, but why not load a variable and update that? Setting a variable wouldn't send a caller-id update to the B leg. What am I missing? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no sdp or contact replacement using externip
On Tue, 16 Jun 2009, Ricardo Martins wrote: Yes Gordon. I'm using nat=yes and I don't have an ALG enabled router/firewall. I used the sip debug output on the asterisk(s) and could see the sdp headers as they were gererated by asterisk, with the wrong (internal) address on it. Asterisk is sending the audio to the correct way, the public IP of client side NAT. But the client is sending it to the wrong address, the private IP of asterisk side NAT. Er, in that case, I can't suggest what might be wrong. All my boxes out in the field are 1.2 though... Gordon Rgrs, Ricardo. Gordon Henderson escreveu: On Tue, 16 Jun 2009, Ricardo Martins wrote: Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. I have dozens of boxes doing it this way. All just work. Have you nat=yes in there too? Also you did port-forward from the router to the box as well, didn't you? Often the router will have a broke SIP ALG which will get in the way too. Turn it off if you can. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] feature keys no longer work after a call has been parked
On Tue, Jun 16, 2009 at 2:23 AM, martin f krafftmadd...@madduck.net wrote: Hey folks, I can park a call with #70 after enabling that feature in features.conf. However, once I retrieve the call from the parking lot, #70 cannot be used to park it again. Worse yet, none of the keys defined in the featuremap work anymore, include blindxfer or automon. Any ideas what may be the problem? Have you set the parkedcallreparking, parkedcalltransfers, and other associated options? -- Jeff Peeler Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Update Caller-ID after Dial()
On Tue, 2009-06-16 at 13:02 +0200, Philipp Kempgen wrote: Can you confirm that currently there is no way to update the caller ID via the manager interface once the B leg is ringing or connected? Correct. Well, at least not with 1.6.0 or 1.6.1 or 1.6.2 branches. Looks like this would be feasible with the functions introduced in https://issues.asterisk.org/view.php?id=8824 ([patch] Remote (called) Party Identification - chan_sip chan_skinny implementation). Yes... that bug number spawned a *lot* of additional work for connected party information (transmission, reception, and updates) that recently went into the trunk of Asterisk. Those features will be available in the 1.6.3 branch of Asterisk, once it has been branched from trunk. I think few people realize just how much work went into getting that feature working in the core of Asterisk, so I'm going to tip my hat to everyone that worked on it and say a big thank you. Such functionality could be desirable in situations when a custom callerid number to name lookup takes more time than I am willing to spend before Dial()ing. It would be desirable in *many* situations, which is why I'm really looking forward to doing more with it in the next few months. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no sdp or contact replacement using externip
Hi all, tks for your time, I could solve the problem on the box that is behind the iptables firewall. I rewrote the rules and the externip is coming with the messages, working with either public and nated uas. I will try further with the asterisk box behind a linksys firewall. The rules I used, just for the record was: For outbound NAT (ppp0 is my external interface of nat/firewall box): iptables -t nat -A POSTROUTING -o ppp0 -j MASQUERADE For port redirection (192.168.1.10 = asterisk internal ip): iptables -t nat -A PREROUTING -i eth0 -p udp -m udp --dport 1:2 -j DNAT --to-destination 192.168.1.10 iptables -t nat -A PREROUTING -i eth0 -p udp -m udp --dport 5060 -j DNAT --to-destination 192.168.1.10 Regards and tks again. If anybody has any issue on this subjet, feel free to ask me. Ricardo. Gordon Henderson escreveu: On Tue, 16 Jun 2009, Ricardo Martins wrote: Yes Gordon. I'm using nat=yes and I don't have an ALG enabled router/firewall. I used the sip debug output on the asterisk(s) and could see the sdp headers as they were gererated by asterisk, with the wrong (internal) address on it. Asterisk is sending the audio to the correct way, the public IP of client side NAT. But the client is sending it to the wrong address, the private IP of asterisk side NAT. Er, in that case, I can't suggest what might be wrong. All my boxes out in the field are 1.2 though... Gordon Rgrs, Ricardo. Gordon Henderson escreveu: On Tue, 16 Jun 2009, Ricardo Martins wrote: Hi all! Do anybody has a full working environment using externip on an asterisk box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp headers info with the externip informed on sip.conf general parameters. I used these two statements: externip=XXX.XXX.XXX.XXX localnet=192.168.200.0/255.255.255.0 Do anybody in list had those dificulties? That's strange because I could not make this work on two diferent instalations! Trying hard to think about what's missing. I have dozens of boxes doing it this way. All "just work". Have you nat=yes in there too? Also you did port-forward from the router to the box as well, didn't you? Often the router will have a broke SIP ALG which will get in the way too. Turn it off if you can. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to CCM
FYI,
Got the Asterisk to Cisco CallManager working over h323. After many
days of trying it was a pretty simple fix.
This is what I had:
[globals]
CISCOTRUNK=H323/callman02
[cisco]
exten = _8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
So if I just write it out without the CISCOTRUNK variable it would look
like this:
exten = _8XXX,1,Dial(H323/callman02/${EXTEN:1...@172.16.200.10:1720)
Turns out all I needed was
exten = _8XXX,1,Dial(H323/${EXTEN:1...@172.16.200.10:1720)
I apparently was wrong in thinking that I needed the h323.conf context
name of my Call Manager configuration (callman02).
Calls are working both ways.
I will put full details in my tutorial at
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html
Jimmy
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy
Ezell
Sent: Thursday, June 11, 2009 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to CCM
Still no luck getting this to work. I have been looking at the
CallManager Logs but so far that is worse then useless. Anyone out
there have any luck connecting Asterisk 1.4 and Cisco CallManager
3.3(5)?
Jimmy Ezell
http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy
Ezell
Sent: Wednesday, June 10, 2009 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk to CCM
As you can see below I am striping off the 8 before it
ever goes to CCM in the extensions.conf file.
exten =
_8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
I have the H323 gateway in CCM configured to use the
same Calling Search Space as my phone extensions.
Jimmy Ezell
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Austin
Sent: Tuesday, June 09, 2009 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk to CCM
Make sure you are stripping the 8 on inbound
calls to that H323 gateway
under CCM and that it has a valid search space
to find your extensions...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy
Ezell
Sent: Tuesday, June 09, 2009 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [asterisk-users] Asterisk to CCM
Hit another problem in my tutorial in converting
over from Cisco CallManager to Asterisk.
I have been following the instructions at :
http://voip-info.capatres.com/wiki/view/Asterisk+Cisco+CallManager+Integ
ration.html on intergrating Asterisk and Cisco CallManager.
I can make calls from CCM to Asterisk phones -
and yes that felt good to get that working.
My problem is that it does not work from the
other direction. I cannot make calls from CCM phones to Asterisk
Phones.
I want to be able to dial 8 and the extension of
the ccm phone.
I am using CCM 3.3.(5) so I do not have the
option to use a SIP turnk because it is not supported. I am also using
h323 instead of ooh323. Not sure if that might make a difference.
In Asterisk console I get:
-- Executing [8...@internal:1]
Dial(SIP/207-08bd64c8, H323/callman02/2...@172.16.200.10:1720) in new
stack
-- Requested transfer capability: 0x00 -
SPEECH
-- Called callman02/2...@172.16.200.10:1720
== Everyone is busy/congested at this time
(1:0/0/1)
This is the contents of my h323.conf file:
=
[general]
port = 1720
bindaddr = 172.17.100.2
disallow=all
[asterisk-users] the correct way to setup a transfer with REFER in SIP
Hi to all excuse me but i don't understand what is the correct configuration needed to setup a transfer with REFER in SIP. I've tried the transfer() application, but i've experienced some problem, i can't reproduce the error in a clear debug environment but randomly the call crash before to be transferred to the final peer. on the wiki (http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer) it is reported as a partial implementation of the REFER functionality. I've tried both atxfer and blindxfer in features.conf but it seems that asterisk make a simple Dial between the two peers. I've also taked a look at ChannelRedirect(channel|[[context|]extension|]priority) but it doesn't seem to be what i need. This is my scenario: A is a SIP Phone registered on the SIP PBX test B is a SIP Phone registered on the SIP PBX test Asterisk is registered on the SIP PBX test with the user C D is a SIP Phone registered on Asterisk. 1) A dial C 2) C (that is Asterisk) execute the dialpan and dial D 3) A and D talks directly as the native bridging is enabled by canreinvite=yes and the codecs are compatible 4) D transfer the call to B What is the configuration needed for the 4th action? My aim is to make a REFER to b...@test and free completely Asterisk. Thanks to all in advance, bye. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Possible Fraud-Mike Low/Zigit/ZonFon/CallCheap
Anyone contacted by a Mike Low of Toronto who does business under a list of names including Zigit, ZonFon, Call Cheap, Kallback King, Cell-0.com, etc. and etc. would be well advised to contact me offline before proceeding to provide services or consulting assistance. -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.ip-pbx.ca www.vpas.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote: Tzafrir Cohen wrote: Duh. Ignore this. You asked about the hardware EC. The hardware EC can be activated regadrdless of the software EC you use. (Not sure exactly how. Anybody?) It's automatic; nothing needs to be specified in /etc/dahdi/system.conf at all. If chan_dahdi is configured to request echo cancellation on the channel, and there is a hardware EC present (and not disabled via module parameters) it will be used. No software EC needs to be configured or even loaded. got told by support to check cat /sys/module/wctdm24xxp/parameters/vpmsupport the thing that is interesting is that dahdi_cfg -vv shows me mg2 (I have mg2 in the /etc/dahdi/system.cfg). Should I just leave echocanceller out fo system.conf ? and dadhi show channel 1 still shows echo cancellation off ? alex -- After all, Europe is America's closest ally. - George W. Bush 02/23/2005 Mainz, Germany signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x
On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote: Alex Samad wrote: some question I have now is when i do a dahdi show channel 1 i get these interesting results Echo Cancellation: 128 taps currently OFF I have a hardware echo can and I have asked for it to be turned on ! that makes sense, I think the confusion here is that the documentation has lots of stuff on the software ec, but not much on the hardware ones for example to use the hardware echo can do I leave echocanceller blank in /etc/dahdi/system.conf ? Why doesn't dahd_cfg -vv show up the hardware ec This is nothing new; the echo canceller on a channel is not enabled unless the channel is in an active call and the call hasn't caused it to be disabled (via tones). If you are looking at a channel that is inactive, this is exactly what you will see. Default law: ulaw I have a alaw:1-4 in the conf file, but it doesn't seem to take That is not valid syntax for /etc/dahdi/system.conf, correct syntax would be 'alaw=1-4'. typo in the email, I actually had alaw=1-4 my last bug bear (maybe bug) is core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) Note the phrase (in microseconds) here. This behavior has changed from Asterisk 1.4, and is documented in the documentation that came with Asterisk 1.6. If you haven't read the CHANGES and UPGRADE files thoroughly, you are spending time trying to understand things (assuming they are problems) that you don't need to spend. very true Something is I think going askew It sounds like you are trying to do a major upgrade without actually taking the time to learn what has changed and what that will require you to do. That's really very important, which is why we spend time writing that documentation in the first place :-) yes that is true, but its a home system, I will do work after falling over the pitfalls at home. -- Just when you think Life's a Bitch, it has puppies. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] missing chan_dahdi.o in debian asterisk 1.4.x
On Wed, Jun 17, 2009 at 07:16:53AM +1000, Alex Samad wrote: On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote: Alex Samad wrote: [snip] Default law: ulaw I have a alaw:1-4 in the conf file, but it doesn't seem to take That is not valid syntax for /etc/dahdi/system.conf, correct syntax would be 'alaw=1-4'. typo in the email, I actually had alaw=1-4 just to follow up I have the above set placed a call to voicemail and did a dahdi show channel 1 the relevant bits Default law: ulaw Echo Cancellation: 256 taps currently ON Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Wed, Jun 17, 2009 at 07:08:10AM +1000, Alex Samad wrote: On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote: Tzafrir Cohen wrote: Duh. Ignore this. You asked about the hardware EC. The hardware EC can be activated regadrdless of the software EC you use. (Not sure exactly how. Anybody?) It's automatic; nothing needs to be specified in /etc/dahdi/system.conf at all. If chan_dahdi is configured to request echo cancellation on the channel, and there is a hardware EC present (and not disabled via module parameters) it will be used. No software EC needs to be configured or even loaded. got told by support to check cat /sys/module/wctdm24xxp/parameters/vpmsupport the thing that is interesting is that dahdi_cfg -vv shows me mg2 (I have mg2 in the /etc/dahdi/system.cfg). Should I just leave echocanceller out fo system.conf ? That's the software EC. It doesn't matter. and dadhi show channel 1 still shows echo cancellation off ? What is the exact value there? Is it at the time of an active call? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help building dahdi for debian
On Wed, Jun 17, 2009 at 01:23:19AM +0300, Tzafrir Cohen wrote: On Wed, Jun 17, 2009 at 07:08:10AM +1000, Alex Samad wrote: On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote: Tzafrir Cohen wrote: Duh. Ignore this. You asked about the hardware EC. The hardware EC can be activated regadrdless of the software EC you use. (Not sure exactly how. Anybody?) It's automatic; nothing needs to be specified in /etc/dahdi/system.conf at all. If chan_dahdi is configured to request echo cancellation on the channel, and there is a hardware EC present (and not disabled via module parameters) it will be used. No software EC needs to be configured or even loaded. got told by support to check cat /sys/module/wctdm24xxp/parameters/vpmsupport the thing that is interesting is that dahdi_cfg -vv shows me mg2 (I have mg2 in the /etc/dahdi/system.cfg). Should I just leave echocanceller out fo system.conf ? That's the software EC. It doesn't matter. ok and dadhi show channel 1 still shows echo cancellation off ? What is the exact value there? Is it at the time of an active call? kevin advised on this, it shows the status at the time of a call. I tried whilst making a call and it was on -- This case has had full analyzation and has been looked at a lot. I understand the emotionality of death penalty cases. - George W. Bush 06/23/2000 Seattle Post-Intelligencer signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.10: core restart on ReceiveFax()
On Tue, Jun 16, 2009 at 10:13 AM, Miguel Molinammol...@millenium.com.co wrote:
sean darcy escribió:
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A fax machines is connected via an SPA 2102 on 173. Any calls from 173
are sent to:
[outbound-fax]
exten = 8447,1,Answer()
exten = 8447,n,GoSub(Capture-Fax,s,1)
exten =_NXXNXX,1,Answer()
exten =_NXXNXX,n,GoSub(DialOut-PSTN,s,1(1${EXTEN}))
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN}))
exten =_91NXXNXX,1,Answer()
exten =_91NXXNXX,n,GoSub(DialOut-PSTN,s,1(${EXTEN:1}))
Actual outbound faxes work correctly. That is, a call from 173 to an
outside fax machine works.
The test faxes go to:
[Capture-Fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
exten = s,n,ReceiveFAX(${FAXFILE}.tif) ;; 1.6 use ReceiveFAX
exten = s,n,Hangup()
When the test fax gets to ReceiveFax() asterisk restarts. Any calls at
the time are lost.
-- Executing [8...@outbound-fax:1] Answer(SIP/173-081d3780, )
in new stack
-- Executing [8...@outbound-fax:2] Gosub(SIP/173-081d3780,
Capture-Fax,s,1) in new stack
-- Executing [...@capture-fax:1] Set(SIP/173-081d3780,
FAXFILE=/var/spool/asterisk/fax/20090612_1710) in new stack
-- Executing [...@capture-fax:2] ReceiveFAX(SIP/173-081d3780,
/var/spool/asterisk/fax/20090612_1710.tif) in new stack
/var/spool/asterisk/fax exists, permissions 777:
ls -l /var/spool/asterisk
total 32
..
drwxrwxrwx 2 root root 4096 2009-05-03 14:21 fax
...
I've set debug and verbose to 20, but no further info.
What am I missing? Anybody have something like this working this working?
sean
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Two thoughts on this:
1. On Exten 8447, why are you using a GoSub instead of a simple Goto?
(Shouldn't pose a problem, just curious)
2. I wouldn't recommend the use of date-time only to form the filename
of the received fax. This was the key issue that was killing asterisk in
my case, when I switched to date-time + uniqueid for example, everything
went fine. That's because it looks like RxFax() or in this case
ReceiveFax() doesn't play well with duplicate (open) filenames, which I
don't know if they occur in your case. It definitely can happen if you
have programmed it and have the capacity to receive simultaneous faxes
at a time. After the change my machine is rock stable, receiving and
sending hundreds of faxes a day with no restarts.
Hope this helps.
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
I've started using GoSub pretty often. No good reason. Just getting
beaten up in class when
I used goto's.
Interesting about the unique id's. But it's not yet my problem. An id
to the minute guarantees uniqueness for my testing.
And I've fixed the problem. I'm now using the new Digium Fax module. That works.
sean
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Re: [asterisk-users] OT: Possible Fraud-Mike Low/Zigit/ZonFon/CallCheap
Yep, heard that a few times before, not just on this list. George Pajari wrote: Anyone contacted by a Mike Low of Toronto who does business under a list of names including Zigit, ZonFon, Call Cheap, Kallback King, Cell-0.com, etc. and etc. would be well advised to contact me offline before proceeding to provide services or consulting assistance. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing LUA
Hello, all. The little bit of reading I've done on lua makes me eager to give it a try. However, when I try to install it (Asterisk 1.6.1.1 on CentOS 5.3), it is not available in menuselect. I have installed lua and lua-devel. I've seen very little about it in my Internet searches. What else must I do so that it installs? Thanks - John Oh, by the way, I'm having a similar problem with speex. I've installed speex and speex-devel but no luck although there does seem to be some known problem with speex_preprocess. I'm assuming I'm going to have to install speex from source if I want to use it with Asterisk. Is that true? Thanks again - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
