[Asterisk-Users] Get sysdate + 5 minutes
Hi,
In my application I want to have the
sysdate + 5 minutes.
I know that the sysdate is in the variable
${DATTIME}
But now I want to now how I get the
sysdate + 5 minutes into a variable?
Does anybody knows the answer?
Kind Regards
Arjan Kroon
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.7.1 crashing my newly built system
The lockup is related to zaptel, i promise :)It's been a while, but I seem to remember Zaptel having some strange interactions with USB in 2.4 kernels. I'd strongly recommend 2.6 On 4/20/06, T.S [EMAIL PROTECTED] wrote: Hello folx!I just started to play with *. I first installed it this past weekend on mySolaris 9 ultra 5 test box. Now I'm attempting to put it on a freshly builtLinux box a mere few hours old. I've installed asterisk 1.2.7.1,libpri-1.2.2, zaptel 1.2.5 and the latest asterisk-sounds 1.2.1.This is running (or installed) on my Slackware 10.2 box running kernel2.4.31.I'm testing with a sip channel (since I have no digital boards), with Xlite 1105x release.The problem I'm having is with the fresh demo install instances. I wastesting out accessing vm and switching back and forth between lines on thexlite client, when I received this message: Apr 19 22:17:00 WARNING[2806]: chan_sip.c:1210 retrans_pkt: Maximum retriesexceeded on transmission [EMAIL PROTECTED] for seqno 49491 (Critical Response)Then immediately my machine became unresponsive and my ssh connectiondropped. I had to reboot it to get back again. I've looked in/var/log/messages to see if I can see the cause, but nothing is there related to asterisk. I don't have this happening at all with my Solaris 9box, which has been running the same tar files since this past weekend. Ireally haven't changed anything from the default except uncomment out the xlite sip user in sip.conf.Can anyone shed some light as to what could be the cause for the error? Thenthe drop in network connection and freezing the machine?Thanks!Terrelle___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modem connection
I have asterisk connected to E1 interface with Digium TE110P. I have Cisco ATA 186 and fax pass thru well. I have tried to establish modem connection (from computer connected to ATA = SIP = * = E1 = Telco = pstn = another modem) and I do connect (at 14,400) but connection end after a minute. How to establish successfully modem connection? Has anybody tried innovaphone IP21 http://www.innovaphone.com/index.php?id=40L=0 They say that fax works with their ATA, maybe modem connection will work also... So, how to establish modem connection? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Get sysdate + 5 minutes
${EPOCH} * Current unix style epoch
Add your 5mins as seconds, and convert if necessary, you could do it
like this in the dialplan to give the same format as ${DATETIME} (which
is deprecated by the way):
${STRFTIME($[${EPOCH} + 300],,%d%m%Y-%H:%M:%S)}
Read doc/README.variables to find out how to do this sort of stuff.
Arjan Kroon wrote:
Hi,
In my application I want to have the sysdate + 5 minutes.
I know that the sysdate is in the variable ${DATTIME}
But now I want to now how I get the sysdate + 5 minutes into a variable?
Doe’s anybody knows the answer?
Kind Regards
Arjan Kroon
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem connection
Tomislav Parčina wrote: I have asterisk connected to E1 interface with Digium TE110P. I have Cisco ATA 186 and fax pass thru well. I have tried to establish modem connection (from computer connected to ATA = SIP = * = E1 = Telco = pstn = another modem) and I do connect (at 14,400) but connection end after a minute. How to establish successfully modem connection? Has anybody tried innovaphone IP21 http://www.innovaphone.com/index.php?id=40L=0 They say that fax works with their ATA, maybe modem connection will work also... Many people say their ATA supports T.38, when it has no T.38 support whatsoever. VoIP boxes are advertised about as honestly as used cars. So, how to establish modem connection? By luck, maybe. The only solution that looks like it should be fairly solid is V.150, and I've only seen that on Cisco boxes so far. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mail issuse when pressing 0
Yeah, I got it a couple of times. Doug Lytle wrote: An outside caller started to leave voice mail. The CLI shows: Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg format: gsm, 0x8295d40 -- x=1, open writing: /var/spool/asterisk/voicemail/sip/4232/INBOX/msg format: wav, 0x829e2c0 -- User cancelled by pressing 0 -- Playing 'vm-saveoper' (language 'en') Later on, the employee tried to retrieve the voice mail, since there was vm indicator on the Polycom. The CLI showed: == Parsing '/var/spool/asterisk/voicemail/sip/4232/INBOX/msg.txt': Found -- Playing 'vm-received' (language 'en') -- Playing 'digits/at' (language 'en') -- Playing 'digits/4' (language 'en') -- Playing 'digits/50' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/p-m' (language 'en') Apr 19 17:02:17 WARNING[19784]: file.c:509 ast_openstream_full: File /var/spool/asterisk/voicemail/sip/4232/INBOX/msg does not exist in any format Apr 19 17:02:17 WARNING[19784]: file.c:821 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/sip/4232/INBOX/msg (format ulaw): No such file or directory I had to manually remove the msg.txt Anybody else see this happening? Asterisk 1.2.7.1 Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement System for a Charity
On 13:40, Thu 20 Apr 06, Douglas Garstang wrote: Does AMP also let you split up each charity so that each only has access to manage their own content? That seems to me to be a pretty big limitation of all the Asterisk management software out there. It's designed to be used by one company to manage their own config, not to be used by many 'organisations' to manage their own data. Kind of like Asterisk being used as a carrier solution rather than a hosted PBX solution. No, that is not possible with AMP/freepbx One of the reasons why I trashed it :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] USB VoIP phone with G729 support
Which USB Phones, come with G729 support? I am looking for one which has the G729 in the software installed on disk itself, so that if the users can use onscreen dialling with headphones if they want. /Obelix ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Get sysdate + 5 minutes
Grab the UNIX timestamp and add 5*60 to it.
In my application I want to have the sysdate + 5 minutes.
I know that the sysdate is in the variable ${DATTIME}
But now I want to now how I get the sysdate + 5 minutes into a variable?
Doe's anybody knows the answer?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with sphinx2
Hi all I install sphinx2 successfully but when i execute sphinx2-server, i have the error below: ad_oss.c(105): Failed to open audio device(/dev/dsp): No such device FATAL_ERROR: server.c, line 476: ad_open() failed What's the matter? I also want to know how i can do in Asterisk to use the sphinx server.Must i write entirely an eagi script? And when it's wrote, which changes can i do in Asterisk or which process does i follow to make sphinx run with Asterisk? Best regards! Serge MESSA OVONO ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queues and the '*' key
On 04/21/06 05:35 Sean Kennedy said the following: I have a vague memory of reading about this somewhere, but searched @ the wiki AND through google aren't turning up anything useful. take a look at http://bugs.digium.com/view.php?id=6897 there's a patch there for 1.2 with another for trunk which has been committed already. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call recording
Have a look at the Dial command, http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial the w and W option allow you to start recording at any time with the *1 keypress. Is there a way to record a call conversation starting in the middle of the call? I know I can recording whole conversation with mixmonitor, but I prefer only recording certain part of the conversation. Thnx. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] some EICON Diva 4BRI questions
Hi! I have some (short) question about the Eicon DIVA (V-)4BRI cards I want to have (short) answered before buying the DIVA card. I know there are several Eicon guys active on the list, thus I ask on the list instead of directly to Eicon so that all other will benefit as well. 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) 3. Is the PIN layout for TE or NT mode? 4. When I change the mode (implies 'YES' for question 1), will this also change the polarity on the connectors or do I have to use a BRI-crossover cable? 5. If a call is bridged from BRI-BRI, is it done directly on the BRI card or via Asterisk 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? Usage in Asterisk: please correct me if I'm wrong: - Communication between the DIVA cards and Asterisk always happens via the CAPI interfaces. - For the Asterisk part I have the choice of chan_misdn or chan_capi(-cm). After reading I come to the conclusion that chan_capi(-cm) should work better. - For the hardware part there are also 2 choices. Either use the Eicon drivers included in 2.6 and divactrl or use the source packages from eicon and the tools included. Here I'm not sure which method is better. Further I do not know how this is related with isdn4linux or other linux ISDN stuff. From: http://www.eicon.com/support/helpweb/slnxen/asterisk.asp 1. Ensure that you do not have the ISDN4Linux driver or the HiSax driver installed. What is the ISDN4Linux driver? Is is something generic in Linux or do they talk about a certain ISDN4Linux driver for Eicon cards? 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. Thanks for clarifications regards Klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meetme scenario
Hi All I want bulding next scenario conferences: Dial number conference roomafter automatic connect 2 users Can any one help me with samples this scenario Thanks Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cubix Softphone + Asterisk 1.2.6
On 20 Apr 2006, at 16:09, Peter Beckman wrote: I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here: http://lists.digium.com/pipermail/asterisk-users/2006-February/ 139240.html I've got a weird problem with both Firefly iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read on the voip-info website that the fix for this with Firefly is to set jitterbuffer to no which I tried but it didn't work. Because the problem is with two IAX softphones I'm not sure whether its a configuration problem with the asterisk server or, by change, the same bug with both softphones. Has anyone else come up against this? Can you change the amount of time between DTMF in Idefisk? Can you modify a config to get Cubix to actually connect to a Dial()ed call? Peter, do you have any packet logs (either ethereal or iax2 debug) of the non-working IAX Dial() I can look at? I may be able to diagnose the problem from that... T. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jingle support - can we test the feature ?
On 20 Apr 2006, at 16:39, Robert Rozman wrote: - Original Message - From: Time Bandit [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 20, 2006 4:18 PM Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ? we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/ There is an Asterisk-plugin that update your status automagically when you're on the phone -- Hi, thanks for pointer. I know for that project, but reading about Jingle, Jabber and Asterisk integration it seems not so interesting for me at the moment... Regards, So what aspect of Jingle, Jabber and Asterisk did you mean in your original post ? Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
Hello Klaus, I will answer your questions In turn: 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? Yes they do. 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) No 3. Is the PIN layout for TE or NT mode? The PIN layout is for TE mode 4. When I change the mode (implies 'YES' for question 1), will this also change the polarity on the connectors or do I have to use a BRI-crossover cable? You would need to use a crossover cable. 5. If a call is bridged from BRI-BRI, is it done directly on the BRI card or via Asterisk It is done on the card. 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? The difference between the Diva Server V-Series and the normal Diva Server cards is that you can only use the V-Series for Voice applications. You can not fax with a V-Series card. If you try and fax with a V-Series card you will get an error. Usage in Asterisk: please correct me if I'm wrong: - Communication between the DIVA cards and Asterisk always happens via the CAPI interfaces. - For the Asterisk part I have the choice of chan_misdn or chan_capi(-cm). After reading I come to the conclusion that chan_capi(-cm) should work better. Yes it is best to use chan-capi-cm as this is developed by the people (Melware) who also develop the Eicon CAPI driver. - For the hardware part there are also 2 choices. Either use the Eicon drivers included in 2.6 and divactrl or use the source packages from eicon and the tools included. Here I'm not sure which method is better. Further I do not know how this is related with isdn4linux or other linux ISDN stuff. From: http://www.eicon.com/support/helpweb/slnxen/asterisk.asp -Eicon can only support it's own drivers - therefore if you require support from Eicon, we would always recommend that you use the Drivers from Eicon. 1. Ensure that you do not have the ISDN4Linux driver or the HiSax driver installed. What is the ISDN4Linux driver? Is is something generic in Linux or do they talk about a certain ISDN4Linux driver for Eicon cards? The ISDN4Linux project was an open source project that allowed you to use the Diva cards. These drivers are not supported by Eicon and we don't have any experience in using them. HiSax is anther open source project for ISDN drivers. 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. This is just a package that contains the capi.h file that is needed to compile the chan_capi-cm driver. Thanks for clarifications regards Klaus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: 21 April 2006 10:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] some EICON Diva 4BRI questions Hi! I have some (short) question about the Eicon DIVA (V-)4BRI cards I want to have (short) answered before buying the DIVA card. I know there are several Eicon guys active on the list, thus I ask on the list instead of directly to Eicon so that all other will benefit as well. 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) 3. Is the PIN layout for TE or NT mode? 4. When I change the mode (implies 'YES' for question 1), will this also change the polarity on the connectors or do I have to use a BRI-crossover cable? 5. If a call is bridged from BRI-BRI, is it done directly on the BRI card or via Asterisk 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? Usage in Asterisk: please correct me if I'm wrong: - Communication between the DIVA cards and Asterisk always happens via the CAPI interfaces. - For the Asterisk part I have the choice of chan_misdn or chan_capi(-cm). After reading I come to the conclusion that chan_capi(-cm) should work better. - For the hardware part there are also 2 choices. Either use the Eicon drivers included in 2.6 and divactrl or use the source packages from eicon and the tools included. Here I'm not sure which method is better. Further I do not know how this is related with isdn4linux or other linux ISDN stuff. From: http://www.eicon.com/support/helpweb/slnxen/asterisk.asp 1. Ensure that you do not have the ISDN4Linux driver or the HiSax driver installed. What is the ISDN4Linux driver? Is is something generic in Linux or do they talk about a certain ISDN4Linux driver for Eicon cards? 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a
[Asterisk-Users] Real-time Database Front-end
Ive had Asterisk working on a test platform really well, but Ive never found a decent web front end, that works in real-time. Ive got a couple of incoming numbers that Id like to have some IVR on (i.e. select this option etc), and then distribute the calls appropriately to various SIP agents, but also in some cases back out to a PSTN/Mobile number. I have this working well on a flat file config, but would like to get it in a db... Ive taken a look at [EMAIL PROTECTED] however I want to install it on a current server, running Fedora Core 4, and the only option I can see is installing it as a new OS or not on FC4. Can anyone provide any advice? Thanks Nunners ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 6.3 unlock/reset?
That only works if the default password is still cisco.. in this case it was not. -- ~Shaun Joseph Rothstein [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] You can usually unlock the phone and then erase the config using the setting sbutton. Push the setting button, nafigate to the bottom of the list, select unlock. Use the keypad to enter the password which is cisco. Undwer network configuraiton there is an erase configuraiton option. Hope this helps. Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, April 21, 2006 11:45, David Waugh said: Hello Klaus, I will answer your questions In turn: 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? Yes they do. 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) No Thus, what is the limitation? TE = slave ? NT = master ? 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? The difference between the Diva Server V-Series and the normal Diva Server cards is that you can only use the V-Series for Voice applications. You can not fax with a V-Series card. If you try and fax with a V-Series card you will get an error. Using spandsp and V-4BRI does not work? What about a pass-through scenario: PSTN -2xBRI- Asterisk -2xBRI- oldPBX Does Asterisk with the V-4BRI allows to pass through fax calls (or other data calls) from the PSTN to the old PBX? 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. This is just a package that contains the capi.h file that is needed to compile the chan_capi-cm driver. btw: libcapi20-dev on debian ;-) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MACRO_RESULT=ABORT
turns out the g option for the macro when using dial doesnt allow both ends to be hung up with setting MACRO_RESULT=ABORT.. My workaround was a bit ghetto, setup a context called hangup-caller and Set(MACRO_RESULT=GOTO:hangup-caller^s^1) jsut fyi.. -- ~Shaun Shaun [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] I have a macro that runs off a dial() and gives the callee a bunch of options... one of them is to disconnect the caller. I read that setting MACRO_RESULT=ABORT would hang up both legs of the call. When i set MACRO_RESULT=ABORT and return to the context it ends up sending the caller to voicemail (the next line in the dialplan/context is voicemail() ). I need it to hangup the call... where am i going wrong? -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 6.3 unlock/reset?
End result was i gave in for a annoying setup... put the phone and a server on a network by it self, setup a dhcp server with a tftp address and flashed the phone to 8.2 which also reset the password back to cisco -- ~Shaun Shaun [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. -- ~Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unicall MFRC2 Problems with BrT.
Hello All, I'm facing problems with Unicall on this scenario : CentOS 4.3 - Running on x86_64 Asterisk 1.2.7.1 Zaptel 1.2.5 When running zttool , shows all Spans OK. But I can't receive and make calls. I tried to change many parameters and still doesn't work. Any clues ? * unicall.conf [channels] language=br context=incoming-pstn usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancellforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both loglevel=255 protocolclass=mfcr2 protocolvariant=br,20,4 protocolend=cpe group=1 callgroup=1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 94-108 channel = 110-124 * zaptel.conf * loadzone=br defaultzone=br span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 span=2,0,0,cas,hdb3 cas=32-46:1101 cas=48-62:1101 span=3,0,0,cas,hdb3 cas=63-77:1101 cas=79-93:1101 span=4,0,0,cas,hdb3 cas=94-108:1101 cas=110-124:1101 * lor error * -- Executing Dial(SIP/1000-1de2, Unicall/g1/40020022|40|Ttr) in new stack Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(1) Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Make call Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0001 - [1/ 1/Idle /Idle ] -- Called g1/40020022 Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - [1/ 40/Seize /Idle ] Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 4 on - [2/ 40/Group I /Idle ] Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 4 off - [1/ 1/Idle /Idle ] Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel gains Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel switching -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/1000-1de2' status is 'CHANUNAVAIL' Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1011 [1/ 1/Idle /Idle ] Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jefferson Carvalho ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom MWI
On Friday 21 April 2006 00:28, Kerry Garrison wrote: Didn't help. Could I be missing something else? In Avi's footsteps, here is my phone.cfg and sip.conf entry. This works for 12 phones. Note that I'm not subscribing to anything on the Polycom; Asterisk sends MWI for the mailbox to the phone when there are messages waiting, and the user can dial '999' to access voicemail. msg msg.bypassInstantMessage=0 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=999 / /msg [211] context=polycom_outgoing type=friend host=dynamic secret=* disallow=all allow=ulaw mailbox=211 vmexten=voicemail dtmfmode=rfc2833 callgroup=1 pickupgroup=1 callerid=Jack 211 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, 21 Apr 2006, Klaus Darilion wrote: Hi! I have some (short) question about the Eicon DIVA (V-)4BRI cards I want to have (short) answered before buying the DIVA card. I know there are several Eicon guys active on the list, thus I ask on the list instead of directly to Eicon so that all other will benefit as well. I try to answer ;-) 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? Yes. 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) I don't know that one. But I can contact Eicon here. 3. Is the PIN layout for TE or NT mode? It is TE PIN layout, you need a crossed cable with 100Ohm termination for NT-mode. 4. When I change the mode (implies 'YES' for question 1), will this also change the polarity on the connectors or do I have to use a BRI-crossover cable? See answer on 3., cross cable is needed. 5. If a call is bridged from BRI-BRI, is it done directly on the BRI card or via Asterisk You can configure this in the capi.conf of chan-capi-cm. If you set bridge=yes, then the b-channels will be bridged on the card, not via Asterisk/CPU. This also works between different cards via Bus-Master DMA. 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? Yes. Usage in Asterisk: please correct me if I'm wrong: - Communication between the DIVA cards and Asterisk always happens via the CAPI interfaces. Correct. - For the Asterisk part I have the choice of chan_misdn or chan_capi(-cm). After reading I come to the conclusion that chan_capi(-cm) should work better. When using Eicon DIVA Server, you can use chan-capi only. chan-misdn is for passive isdn cards like the DIVA client cards. - For the hardware part there are also 2 choices. Either use the Eicon drivers included in 2.6 and divactrl or use the source packages from eicon and the tools included. Here I'm not sure which method is better. Further I do not know how this is related with isdn4linux or other linux ISDN stuff. From: http://www.eicon.com/support/helpweb/slnxen/asterisk.asp The in-kernel driver in 2.6 is the so called v2 driver from Melware. This driver works very good. But the driver from Eicons sourceRPM (Melware calls it v3) is the newer one with many more features, more supported cards, newer firmware... (like RTP support which is used with newer chan-capi-cm as well). 1. Ensure that you do not have the ISDN4Linux driver or the HiSax driver installed. What is the ISDN4Linux driver? Is is something generic in Linux or do they talk about a certain ISDN4Linux driver for Eicon cards? ISDN4Linux is the 'old' ISDN core of the Linux kernel. It is still there, but not developed any more. CAPI is used more and more. The HiSax driver as well as older Eicon driver in kernel uses this core. HiSax is the old driver for passive ISDN cards, mISDN is the new one here. With CAPI and Eicon DIVA Server you don't need ISDN4Linux, but it should not harm if installed in parallel, because CAPI can co-exist with it. We (Melware) do provide a special driver where the new Eicon drivers can be used with ISDN4Linux as well. E.g. when you want to use the AT-emulator (ttyI) interfaces of ISDN4Linux with the Eicon DIVA Server cards. But Eicon does provide an own special tty-AT-emulator as well. All these interfaces (CAPI, ISDN4Linux, tty-AT) can be used at the same time if wanted. 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. The isdn4k-utils (isdn for kernel utilities) contain the CAPI library (libcapi20) which is needed for chan-capi to compile/use. That is the reason for that package. Just make sure that you have the libcapi20.* and the header files of it (some distributions provide the header files via a special -devel package only). Since I have created a special version of that libcapi20 to support remote-CAPI via TCP (ISDN Hardware in one maschine, Application in another), Melware will provide more packages via chan-capi.org / melware.org soon. If you have further questions, please don't hesitate to ask. Armin -- Cytronics Melware ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AAH or Fedora an Asterisk by sources
Hello list users, I come to you in order to ask you your best recommendation for a large scale production Server, which Hill be your best recommendation between [EMAIL PROTECTED] and an installation from scratch with Fedora Core 4 and asterisk compiled by sources?? Thanks in advance, have fun! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, 21 Apr 2006, Klaus Darilion wrote: On Fri, April 21, 2006 11:45, David Waugh said: 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? The difference between the Diva Server V-Series and the normal Diva Server cards is that you can only use the V-Series for Voice applications. You can not fax with a V-Series card. If you try and fax with a V-Series card you will get an error. Using spandsp and V-4BRI does not work? That will work. It's just that the on-board fax capabilities won't work, but any other software fax will work like with other cards. What about a pass-through scenario: PSTN -2xBRI- Asterisk -2xBRI- oldPBX No Problem! I have this running with 2 4BRI cards: PSTN -4xBRI- Asterisk/OpenPBX -4xBRI NT-mode- old PBX Does Asterisk with the V-4BRI allows to pass through fax calls (or other data calls) from the PSTN to the old PBX? Yes, but bridge=yes should be used to pass trough without Asterisk/CPU. 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. This is just a package that contains the capi.h file that is needed to compile the chan_capi-cm driver. btw: libcapi20-dev on debian ;-) exactly! Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
Hello Klaus, Normally, Diva Server adapters are operated as terminal equipment. In this case, they derive their timing from the signal received from the NT, for example PSTN or PBX, and use this derived timing to synchronize their transmitted signal. If you use the Diva Server adapter as network termination, it generates the timing from which the terminal equipment derives its timing and synchronization. Thus TE=Slave And NT=Master I haven't tried spandsp and a V-Series card as the card was not designed for fax applications. As a result it has never been tested. If you want to fax with the card then we would recommend a normal Diva Server card. It would probably work in a pass through scenario as just audio is being relayed by the card. However, I have not tested this and could not comment if this will work with a V-BRI card. Bascially if you are looking to do pure voice, then use a V-Series card. If you want to do anything more than voice, then use a normal Diva Server card. Kind regards David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: 21 April 2006 11:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions On Fri, April 21, 2006 11:45, David Waugh said: Hello Klaus, I will answer your questions In turn: 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? Yes they do. 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) No Thus, what is the limitation? TE = slave ? NT = master ? 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? The difference between the Diva Server V-Series and the normal Diva Server cards is that you can only use the V-Series for Voice applications. You can not fax with a V-Series card. If you try and fax with a V-Series card you will get an error. Using spandsp and V-4BRI does not work? What about a pass-through scenario: PSTN -2xBRI- Asterisk -2xBRI- oldPBX Does Asterisk with the V-4BRI allows to pass through fax calls (or other data calls) from the PSTN to the old PBX? 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. This is just a package that contains the capi.h file that is needed to compile the chan_capi-cm driver. btw: libcapi20-dev on debian ;-) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] record_in / record_out configuration parameters
Hi all, having performance problems with various SIP-Phones, the manufacturer adviced us to add these parameters in sip.conf - unfortunately, neither one of us has an idea what these are supposed to do. I've seen various configuration files (sip.conf, iax.conf) posted on the net or this list using said paramters, but they seem to completely lack documentation (or is it just me?). Grepping for them in the asterisk sources (and those of related packages) didn't show up anything either. We're running asterisk 1.2, just in case they're old parameters which have been removed or more recently added ones... Could someone in the know give a short explanation of what these actually do? Thanx in advance, Flo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AAH or Fedora an Asterisk by sources
Install from sources. Run only Asterisk on the production machine, no extra processes. If you must have databases, GUI, AGI and ... put those on another machine and call them across the network. Also, I am not sure how much it helps, but eliminate asterisk modules you will not use by using noload= in modules.conf. chmod 444 the files that you do not want to overwrite on the production box (zapata.conf, zaptel.conf, xxx_addtional, ...) If you like [EMAIL PROTECTED], install it on a test machine. Make all of your configs there and then copy the .confs to the production server (I use WinSCP to copy and also keep a set of copies in Source Safe.) Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Fri 4/21/2006 6:28 AM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] AAH or Fedora an Asterisk by sources Hello list users, I come to you in order to ask you your best recommendation for a large scale production Server, which Hill be your best recommendation between [EMAIL PROTECTED] and an installation from scratch with Fedora Core 4 and asterisk compiled by sources?? Thanks in advance, have fun! winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on Red Hat AS 4?
Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swift application??
Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Red Hat AS 4?
Red Hat AS 4 is the same as CentOS4X (with the Red Hat references stripped) and Asterisk works just fine on it. -Original Message- From: Mimmus [mailto:[EMAIL PROTECTED] Sent: Fri 4/21/2006 6:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: Subject: [Asterisk-Users] Asterisk on Red Hat AS 4? Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swiftapplication??
Type swift at the command line so you can see the -options. Then modify the line to use the correct switch and specify the name of the voice you want to use. Thanks, Steve -Original Message- From: Pimjai Wesnarat [mailto:[EMAIL PROTECTED] Sent: Fri 4/21/2006 6:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swiftapplication?? Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cubix Softphone + Asterisk 1.2.6
Cubix has always crashed on me while using moderately. Nice looking phone but not stable. Idefisk works great. On 20 Apr 2006, at 16:09, Peter Beckman wrote: I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here: http://lists.digium.com/pipermail/asterisk-users/2006-February/ 139240.html I've got a weird problem with both Firefly iaxLite (both IAX softphones). They don't seem to stop ringing when an incoming call is make to them. If the call is answered the conversation starts both ways but the ringing sound still keeps going and the softphones keep displaying that a call is coming in (but they do not display that the call is answered). I read on the voip-info website that the fix for this with Firefly is to set jitterbuffer to no which I tried but it didn't work. Because the problem is with two IAX softphones I'm not sure whether its a configuration problem with the asterisk server or, by change, the same bug with both softphones. Has anyone else come up against this? Can you change the amount of time between DTMF in Idefisk? Can you modify a config to get Cubix to actually connect to a Dial()ed call? Peter, do you have any packet logs (either ethereal or iax2 debug) of the non-working IAX Dial() I can look at? I may be able to diagnose the problem from that... T. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Redirecting to another service/server
Did you get an answer to this? I am interested in SIP to SIP calls on other networks thereby by-passing the pstn. -- Original message -- From: Nick Hoffman [EMAIL PROTECTED] Hi guys. Without having a FWD account, can Asterisk redirect calls to FWD? For instance, an extension behind Asterisk dials 99751234, and Asterisk says "that starts with 99. let's strip off the 99 and call 751234 at FWD, IE: sip:[EMAIL PROTECTED]:5060". Is that possible, or would services such as FWD reject the call since the device making the call (Asterisk) hasn't registered? Thanks! -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated wi th it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, April 21, 2006 12:33, David Waugh said: Hello Klaus, ... I haven't tried spandsp and a V-Series card as the card was not designed for fax applications. As a result it has never been tested. If you want to fax with the card then we would recommend a normal Diva Server card. It would probably work in a pass through scenario as just audio is being relayed by the card. However, I have not tested this and could not comment if this will work with a V-BRI card. Bascially if you are looking to do pure voice, then use a V-Series card. If you want to do anything more than voice, then use a normal Diva Server card. Hi David! You never know which kind of voice (human, modem, fax) will pass through the B channel. I guess handling pass-trough modem calls should work, if they card will disable the echo canceller on detection of the special tones used in modem conversations. Do they? regards klaus Kind regards David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: 21 April 2006 11:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions On Fri, April 21, 2006 11:45, David Waugh said: Hello Klaus, I will answer your questions In turn: 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? Yes they do. 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) No Thus, what is the limitation? TE = slave ? NT = master ? 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? The difference between the Diva Server V-Series and the normal Diva Server cards is that you can only use the V-Series for Voice applications. You can not fax with a V-Series card. If you try and fax with a V-Series card you will get an error. Using spandsp and V-4BRI does not work? What about a pass-through scenario: PSTN -2xBRI- Asterisk -2xBRI- oldPBX Does Asterisk with the V-4BRI allows to pass through fax calls (or other data calls) from the PSTN to the old PBX? 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. This is just a package that contains the capi.h file that is needed to compile the chan_capi-cm driver. btw: libcapi20-dev on debian ;-) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Airspan / Arelnet GW and Asterisk
Hi allHas anyone seen this kind of messages : Apr 21 12:04:12 NOTICE[89928]: chan_sip.c:3449 process_sdp: Content is 'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp'I get this using a priorietary (Airspan's prime, ex-arelnet) E1 gateway with asterisk. It seems like the SIP protocol they're using is somehow weird...Is there a configuration-level solution or should i patch asterisk so that it ignores the trailing multipart content of the INVITE request ?(for the record, the end of the request looks like :a=rtpmap:0 PCMU/8000a=ptime:40a=rtpmap:17 T38/8000a=ptime:40a=ptime:20--unique-boundary-1Content-Type: application/x-Arelnet;Object=OrigTrunkLNo=1--unique-boundary-1- (11 headers 26 lines)---) -- Michel Luczak[EMAIL PROTECTED]***Internet Email Confidentiality Footer**Privileged/Confidential Information may be contained in this message.If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not copy or deliver this message to anyone.In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer does not consent to Internet email for messages of this kind. Opinions, conclusions and other information in this message that do not relate to the official business of my firm shall be understood as neither given nor endorsed by it.*** ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Hi Armin! Thanks for the detailed answers and isdn for Linux basics. I will take the opportunity to ask some more questions :-) On Fri, April 21, 2006 12:24, Armin Schindler said: On Fri, 21 Apr 2006, Klaus Darilion wrote: 3. Is the PIN layout for TE or NT mode? It is TE PIN layout, you need a crossed cable with 100Ohm termination for NT-mode. Why do I need a cable with 100Ohm termination? Shouldn't be the termination inside the DIVA Server? Until now (with quadbri and other isdn card) I only used CAT5 cables with BRI-crossover PIN layout. No resistors. Can you please explain this a little bit more or give me links to the wiring basics? 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? Yes. This means all kind of B-channel data (human voice, modem, fax, data ...) can be handled in pass-through also with the V-4BRI card? - For the hardware part there are also 2 choices. Either use the Eicon drivers included in 2.6 and divactrl or use the source packages from eicon and the tools included. Here I'm not sure which method is better. Further I do not know how this is related with isdn4linux or other linux ISDN stuff. From: http://www.eicon.com/support/helpweb/slnxen/asterisk.asp The in-kernel driver in 2.6 is the so called v2 driver from Melware. This driver works very good. But the driver from Eicons sourceRPM (Melware calls it v3) is the newer one with many more features, more supported cards, newer firmware... (like RTP support which is used with newer chan-capi-cm as well). On the Eicon homepage I find to source packages: version 7.7 (they call ist stable) and version 8.0 (they call it beta). To which one of these do you refer with v3? Or is there another version somewhere hidden on the homepage? What is meant with newer firmware? Is there a new firmware available which must be flashed into the cards? Is the new v3 ready to use in production environment? thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Real-time Database Front-end
James Nunnerley wrote: I’ve had Asterisk working on a test platform really well, but I’ve never found a decent web front end, that works in real-time. I’ve got a couple of incoming numbers that I’d like to have some IVR on (i.e. select this option etc), and then distribute the calls appropriately to various SIP agents, but also in some cases back out to a PSTN/Mobile number. I have this working well on a flat file config, but would like to get it in a db... I’ve taken a look at [EMAIL PROTECTED] however I want to install it on a current server, running Fedora Core 4, and the only option I can see is installing it as a new OS – or not on FC4. Can anyone provide any advice? http://asteriskathome.sourceforge.net/handbook/index.html Section 2.3 says... mkdir /var/aah_load cp asteriskathome-1.5.tar.gz /var/aah_load cd /var/aah_load tar xvfz asteriskathome-1.5.tar.gz ./install.sh Change the 1.5 to whatever is current; should work. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
Hi! I've forgotten to ask an important question: Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode? thanks klaus btw: we should collect QA somewhere on a Wiki. On Fri, April 21, 2006 12:33, David Waugh said: Hello Klaus, Normally, Diva Server adapters are operated as terminal equipment. In this case, they derive their timing from the signal received from the NT, for example PSTN or PBX, and use this derived timing to synchronize their transmitted signal. If you use the Diva Server adapter as network termination, it generates the timing from which the terminal equipment derives its timing and synchronization. Thus TE=Slave And NT=Master I haven't tried spandsp and a V-Series card as the card was not designed for fax applications. As a result it has never been tested. If you want to fax with the card then we would recommend a normal Diva Server card. It would probably work in a pass through scenario as just audio is being relayed by the card. However, I have not tested this and could not comment if this will work with a V-BRI card. Bascially if you are looking to do pure voice, then use a V-Series card. If you want to do anything more than voice, then use a normal Diva Server card. Kind regards David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: 21 April 2006 11:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions On Fri, April 21, 2006 11:45, David Waugh said: Hello Klaus, I will answer your questions In turn: 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? Yes they do. 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) No Thus, what is the limitation? TE = slave ? NT = master ? 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? The difference between the Diva Server V-Series and the normal Diva Server cards is that you can only use the V-Series for Voice applications. You can not fax with a V-Series card. If you try and fax with a V-Series card you will get an error. Using spandsp and V-4BRI does not work? What about a pass-through scenario: PSTN -2xBRI- Asterisk -2xBRI- oldPBX Does Asterisk with the V-4BRI allows to pass through fax calls (or other data calls) from the PSTN to the old PBX? 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. This is just a package that contains the capi.h file that is needed to compile the chan_capi-cm driver. btw: libcapi20-dev on debian ;-) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom MWI
Ohh yeah good point. I had a similar issue when I started using FreePBX and it didn't fill out the mailbox field automatically. Once I added the [EMAIL PROTECTED] there the MWI started working as well. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Friday, April 21, 2006 12:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom MWI Kerry Garrison wrote: Didn't help. Could I be missing something else? My phone.cfg looks like this: mwi msg.mwi.1.subscribe=300 msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97/ And sip.conf for extension 300: [300] username=300 type=friend secret=*** record_out=Adhoc record_in=Adhoc qualify=no port=5060 pickupgroup=1 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all context=from-internal canreinvite=no callgroup=1 callerid=Polycom IP501 300 allow=alaw allow=g729 Mine works fine, so I hope that helps. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, 21 Apr 2006, Klaus Darilion wrote: On Fri, April 21, 2006 12:33, David Waugh said: Hello Klaus, ... I haven't tried spandsp and a V-Series card as the card was not designed for fax applications. As a result it has never been tested. If you want to fax with the card then we would recommend a normal Diva Server card. It would probably work in a pass through scenario as just audio is being relayed by the card. However, I have not tested this and could not comment if this will work with a V-BRI card. Bascially if you are looking to do pure voice, then use a V-Series card. If you want to do anything more than voice, then use a normal Diva Server card. Hi David! You never know which kind of voice (human, modem, fax) will pass through the B channel. I guess handling pass-trough modem calls should work, if they card will disable the echo canceller on detection of the special tones used in modem conversations. Do they? The echo-chanceler can do that, but chan-capi disables echo-cancel on bridge anyway. So on bridge no data is modified. Also the automatic disable of echocancel on e.g. a fax-tone when using a software fax can be configured as option in capi.conf of chan-capi. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Armin Schindler wrote: Using spandsp and V-4BRI does not work? That will work. It's just that the on-board fax capabilities won't work, but any other software fax will work like with other cards. Just a note that I've never managed to get this to work on my V-4BRI cards: If I attempt to use SpanDSP to send or receive a fax, Asterisk will crash. This happens on multiple servers, so now I don't even bother compiling SpanDSP support onto my BRI-only Asterisk servers. If anyone knows how to actually get this working, I'm all ears. cYa, Avi -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, 21 Apr 2006, Klaus Darilion wrote: Hi! I've forgotten to ask an important question: Does Diva Server V-4BRI with Asterisk support BRI P2P and P2MP mode? Yes, and each port can be configured separately. thanks klaus btw: we should collect QA somewhere on a Wiki. Yes. For that purpose Melware has activated melware.org Armin On Fri, April 21, 2006 12:33, David Waugh said: Hello Klaus, Normally, Diva Server adapters are operated as terminal equipment. In this case, they derive their timing from the signal received from the NT, for example PSTN or PBX, and use this derived timing to synchronize their transmitted signal. If you use the Diva Server adapter as network termination, it generates the timing from which the terminal equipment derives its timing and synchronization. Thus TE=Slave And NT=Master I haven't tried spandsp and a V-Series card as the card was not designed for fax applications. As a result it has never been tested. If you want to fax with the card then we would recommend a normal Diva Server card. It would probably work in a pass through scenario as just audio is being relayed by the card. However, I have not tested this and could not comment if this will work with a V-BRI card. Bascially if you are looking to do pure voice, then use a V-Series card. If you want to do anything more than voice, then use a normal Diva Server card. Kind regards David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: 21 April 2006 11:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] some EICON Diva 4BRI questions On Fri, April 21, 2006 11:45, David Waugh said: Hello Klaus, I will answer your questions In turn: 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? Yes they do. 2. Can the clock (master/slave) be configured independent from the mode (TE/NT) No Thus, what is the limitation? TE = slave ? NT = master ? 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? The difference between the Diva Server V-Series and the normal Diva Server cards is that you can only use the V-Series for Voice applications. You can not fax with a V-Series card. If you try and fax with a V-Series card you will get an error. Using spandsp and V-4BRI does not work? What about a pass-through scenario: PSTN -2xBRI- Asterisk -2xBRI- oldPBX Does Asterisk with the V-4BRI allows to pass through fax calls (or other data calls) from the PSTN to the old PBX? 2. ...install the isdn4k-utils-devel package What is isdn4k ? I'm a little bit confused about what I have to install. This is just a package that contains the capi.h file that is needed to compile the chan_capi-cm driver. btw: libcapi20-dev on debian ;-) regards klaus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, 21 Apr 2006, Avi Miller wrote: Armin Schindler wrote: Using spandsp and V-4BRI does not work? That will work. It's just that the on-board fax capabilities won't work, but any other software fax will work like with other cards. Just a note that I've never managed to get this to work on my V-4BRI cards: If I attempt to use SpanDSP to send or receive a fax, Asterisk will crash. This happens on multiple servers, so now I don't even bother compiling SpanDSP support onto my BRI-only Asterisk servers. If anyone knows how to actually get this working, I'm all ears. I don't know much about spandsp, but a crash should not happen. Since Asterisk crashes, the problem is not the divas driver. It could be chan-capi, but I don't why this would make a difference, because for chan-capi it is voice data just like with any other voice connection. Can you provide a backtrace to track this problem down? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to select Ceptral's Voice in Asterisk's Swift application??
Quoting Pimjai Wesnarat [EMAIL PROTECTED]: Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? [cepstral-demo] exten = s,1,Answer exten = s,n,wait(1) exten = s,n,Cepstral(voice name=DuchessHello and welcome to the world of text to speech using Cepstral. My name is Duchess./voice) exten = s,n,Cepstral(voice name=WalterHello and welcome to the world of text to speech using Cepstral. My name is Walter./voice) exten = s,n,Cepstral(voice name=ShoutyHello and welcome to the world of text to speech using Cepstral. My name is Shouty./voice) exten = s,n,Cepstral(voice name=WilliamHello and welcome to the world of text to speech using Cepstral. My name is William./voice) exten = s,n,Cepstral(voice name=WhisperyHello and welcome to the world of text to speech using Cepstral. My name is Whispery./voice) exten = s,n,Cepstral(voice name=RobinHello and welcome to the world of text to speech using Cepstral. My name is Robin./voice) exten = s,n,Cepstral(voice name=LindaHello and welcome to the world of text to speech using Cepstral. My name is Linda./voice) exten = s,n,Cepstral(voice name=EmilyHello and welcome to the world of text to speech using Cepstral. My name is Emily./voice) exten = s,n,Cepstral(voice name=DianeHello and welcome to the world of text to speech using Cepstral. My name is Diane./voice) exten = s,n,Cepstral(voice name=DavidHello and welcome to the world of text to speech using Cepstral. My name is David./voice) exten = s,n,Cepstral(voice name=DuncanHello and welcome to the world of text to speech using Cepstral. My name is Duncan./voice) exten = s,n,Cepstral(voice name=DamienHello and welcome to the world of text to speech using Cepstral. My name is Damien./voice) exten = s,n,Cepstral(voice name=CallieHello and welcome to the world of text to speech using Cepstral. My name is Callie./voice) exten = s,n,Cepstral(voice name=DogHello and welcome to the world of text to speech using Cepstral. My name is Dog./voice) exten = s,n,Cepstral(voice name=AmyHello and welcome to the world of text to speech using Cepstral. My name is Amy./voice) This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
Hi Klaus, Thanks for the detailed answers and isdn for Linux basics. I will take the opportunity to ask some more questions :-) On Fri, April 21, 2006 12:24, Armin Schindler said: On Fri, 21 Apr 2006, Klaus Darilion wrote: 3. Is the PIN layout for TE or NT mode? It is TE PIN layout, you need a crossed cable with 100Ohm termination for NT-mode. Why do I need a cable with 100Ohm termination? Shouldn't be the termination inside the DIVA Server? Until now (with quadbri and other isdn card) I only used CAT5 cables with BRI-crossover PIN layout. No resistors. Can you please explain this a little bit more or give me links to the wiring basics? I must admit that I don't really know that. Maybe the quadbri has this termination automatically on board. Since the Eicon DIVA Server card has basically a TE port, the termination is necessary. I always use the 100Ohm termination in my NT-cross-cables. Here is a short description on Melware Wiki: http://www.melware.org/BriCrossCable 6. Difference between V-4BRI and 4BRI: As far as I understand the 4BRI is the better (more expensive) card which also offers FAX on/offramp. Nevertheless I can use V-4BRI for faxing when using spandsp. Both cards do support onboard echo cancellation. Are this assumptions correct? Yes. This means all kind of B-channel data (human voice, modem, fax, data ...) can be handled in pass-through also with the V-4BRI card? Yes. In bridge mode (CAPI calls that Line-Interconnect) the b-channels are connected together, no matter what kind of data is going through. - For the hardware part there are also 2 choices. Either use the Eicon drivers included in 2.6 and divactrl or use the source packages from eicon and the tools included. Here I'm not sure which method is better. Further I do not know how this is related with isdn4linux or other linux ISDN stuff. From: http://www.eicon.com/support/helpweb/slnxen/asterisk.asp The in-kernel driver in 2.6 is the so called v2 driver from Melware. This driver works very good. But the driver from Eicons sourceRPM (Melware calls it v3) is the newer one with many more features, more supported cards, newer firmware... (like RTP support which is used with newer chan-capi-cm as well). On the Eicon homepage I find to source packages: version 7.7 (they call ist stable) and version 8.0 (they call it beta). To which one of these do you refer with v3? Or is there another version somewhere hidden on the homepage? Melware's v3 references both. v3 is just the version of the drivers/firmware compared with the v2 which is the status of the driver in kernel 2.6. But please don't be confused with this v3 naming. The Eicon source package (7.7 or 8.0) is very good. Melware will create a v3 driver package out of that driver status including some open-source community changes/add-ons. But this is not finished yet. What is meant with newer firmware? Is there a new firmware available which must be flashed into the cards? Not flashed, but loaded. When the Eicon DIVA Server cards are started, the tool divactrl will activate the card by loading the firmware automatically and configuring it (the ports) according to settings. New firmware means: new features has been added! Remember, these cards are active cards. A system is running on that card handling the ISDN protocol (the hosts CPU/the driver does not need to care about that) as well as the b-channel data processing for echo-cancel, fax, modem, RTP, etc. Is the new v3 ready to use in production environment? As written above the Melware's v3 is not finished yet, but the Eicons source package is definately ready for that (we have it running in many production systems with different applications). Also, if you find any problem or missing feature, we can help you very fast since we work very closely with Eicon. And even there is a change in the firmware needed, Eicons response time is excellent. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Modem connection
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... By luck, maybe. The only solution that looks like it should be fairly solid is V.150, and I've only seen that on Cisco boxes so far. Hi Steve! Can you tell me more about Cisco box that are you talking about? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash Panel / Queue Slots
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, is there any way to make the Flash Operator Panel show which agents are logged in in a specific queue? (both static and dynamic agents) I've played around with the queue / queue agents settings from the Flash Panel documentation (http://www.asternic.org). The way it is described there, I could only make the Flash panel show that a queue 8in general) received a call from a specific extension. - -- Thomas Broda, Systemadministration Frankfurt FIRSTGATE AG,Im MediaPark 5, 50670 Koeln Telefon: +49 (0) 2 21 / 45 45-747 Telefax: +49 (0) 2 21 / 45 45-710 Internet: http://www.firstgate.de -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFESOZZulxz1xno2o4RApgqAJ99LdLeExyRPPCOTMkl/qGOWYW7XwCdFRRQ Nsg9vqxFtilaPaeqwSn1v3I= =YV2A -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom MWI
Thanks a ton!! When using Extensions mode (the default) this would be: [EMAIL PROTECTED] When Using Users and Devices mode this would be: [EMAIL PROTECTED] Thanks for the guidance there, this has been driving me nuts. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Friday, April 21, 2006 5:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom MWI Ohh yeah good point. I had a similar issue when I started using FreePBX and it didn't fill out the mailbox field automatically. Once I added the [EMAIL PROTECTED] there the MWI started working as well. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller Sent: Friday, April 21, 2006 12:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom MWI Kerry Garrison wrote: Didn't help. Could I be missing something else? My phone.cfg looks like this: mwi msg.mwi.1.subscribe=300 msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97/ And sip.conf for extension 300: [300] username=300 type=friend secret=*** record_out=Adhoc record_in=Adhoc qualify=no port=5060 pickupgroup=1 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all context=from-internal canreinvite=no callgroup=1 callerid=Polycom IP501 300 allow=alaw allow=g729 Mine works fine, so I hope that helps. :) -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom MWI
Try specifing [EMAIL PROTECTED] I know their have been some changes with the implicit defining of the voicemail groupsthat may have something to do with it... I didn't have to do anything special for my polycoms. Sean On Fri, 2006-04-21 at 06:17 -0400, Andrew Kohlsmith wrote: On Friday 21 April 2006 00:28, Kerry Garrison wrote: Didn't help. Could I be missing something else? In Avi's footsteps, here is my phone.cfg and sip.conf entry. This works for 12 phones. Note that I'm not subscribing to anything on the Polycom; Asterisk sends MWI for the mailbox to the phone when there are messages waiting, and the user can dial '999' to access voicemail. msg msg.bypassInstantMessage=0 mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=999 / /msg [211] context=polycom_outgoing type=friend host=dynamic secret=* disallow=all allow=ulaw mailbox=211 vmexten=voicemail dtmfmode=rfc2833 callgroup=1 pickupgroup=1 callerid=Jack 211 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unicall MFRC2 Problems with BrT.
A couple of weeks ago, libmfcr2 has a small error in the tone signaling for the call setup, that was fixed 2 weeks ago or so, please, wich version of libmfcr2 are you using? if you dont know try upgrading to the latest version. Im pretty much sure that you have the very same problem we had. Regards On 4/21/06, Jefferson Carvalho [EMAIL PROTECTED] wrote: Hello All, I'm facing problems with Unicall on this scenario : CentOS 4.3 - Running on x86_64 Asterisk 1.2.7.1 Zaptel 1.2.5 When running zttool , shows all Spans OK. But I can't receive and make calls. I tried to change many parameters and still doesn't work. Any clues ? * unicall.conf [channels] language=br context=incoming-pstn usecallerid=yes hidecallerid=no immediate=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancellforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 faxdetect=both loglevel=255 protocolclass=mfcr2 protocolvariant=br,20,4 protocolend=cpe group=1 callgroup=1 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 94-108 channel = 110-124 * zaptel.conf * loadzone=br defaultzone=br span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 span=2,0,0,cas,hdb3 cas=32-46:1101 cas=48-62:1101 span=3,0,0,cas,hdb3 cas=63-77:1101 cas=79-93:1101 span=4,0,0,cas,hdb3 cas=94-108:1101 cas=110-124:1101 * lor error * -- Executing Dial(SIP/1000-1de2, Unicall/g1/40020022|40|Ttr) in new stack Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(1) Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Make call Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0001 - [1/ 1/Idle /Idle ] -- Called g1/40020022 Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - [1/ 40/Seize /Idle ] Apr 20 19:13:57 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 4 on - [2/ 40/Group I /Idle ] Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 R2 prot. err. [2/ 40/Group I /DNIS ] cause 32769 - T1 timed out Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 4 off - [1/ 1/Idle /Idle ] Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Protocol failure -- Unicall/1 protocol error. Cause 32769 Apr 20 19:14:02 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel gains Apr 20 19:14:03 WARNING[30676]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel switching -- Hungup 'UniCall/1-1' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/1000-1de2' status is 'CHANUNAVAIL' Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1011 [1/ 1/Idle /Idle ] Apr 20 19:14:03 WARNING[30664]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/ 1/Idle /Idle ] Jefferson Carvalho ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with TE205
Hello,
I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my
jumper set (i.e closed to use the E1 facility.)
but when i connect the E1 from my telco the LED on the TDM card is green
and also when i look in zttool the status are ok.
when i try to place a call out I get the following:
-- Accepting AUTHENTICATED call from 196.1.178.172:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (g729|gsm),
priority = mine
-- Executing Dial(IAX2/233066391-4, Zap/g5/8105156,30) in new
stack
Apr 21 13:50:43 NOTICE[1565]: app_dial.c:1011 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
= Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'IAX2/233066391-6'
-- Accepting AUTHENTICATED call from 196.1.178.172:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (g729|gsm),
priority = mine
-- Executing Dial(IAX2/233066391-4, Zap/g5/8105156,30) in new
stack
Apr 21 14:28:52 NOTICE[1095]: app_dial.c:1011 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
pbx23*CLI
My config files:
zaptel.conf
===
span=1,1,0,ccs,hdb3
span=2,1,0,ccs,hdb3
bchan=1-15,17-30
dchan=16
bchan=31-45,47-60
dchan=46
loadzone = us
zapata.conf
group=5
context=outsideserver
;context= incoming
switchtype=euroisdn
pridialplan=national
priindication=inband
signalling=pri_cpe
resetinterval=60
overlapdial=yes
callprogress=yes
callerid=22004488
;channel=1-15,17-30
bchan=1-15,17-31
dchan=16
txgain=3.0
rxgain=3.0
group=7
context=weareout
;context= incoming
switchtype=euroisdn
;context= incoming
switchtype=euroisdn
pridialplan=national
priindication=inband
signalling=pri_cpe
resetinterval=60
overlapdial=yes
callprogress=yes
callerid=155152
;channel=31-45
;channel=47-61
bchan=31-45,47-60
dchan=46
txgain=3.0
rxgain=3.0
idledial=1551520
[EMAIL PROTECTED]
minunused=2
minidle=1
and this is my dialplan.trying to call out on group 5
extensions.conf
==
[outsideserver]
include= default
include= national
include= local
[weareout]
exten = 1551520,1,Answer
..
..
[national]
exten =_02X.,1,Dial,Zap/g5/${EXTEN:2},30
exten = _0802X.,1,Dial,Zap/g5/${EXTEN},30
exten = _080[36]X.,1,Dial,Zap/g5/${EXTEN},30
;exten = _0804X.,1,Dial,Zap/g5/${EXTEN},30
could this be a hardware problem? or is the TDM card defected?
Please anyone ever had this problem i will like to share how it was sorted
out.
regards
==
Olaifa Augustine
General Data Engineering Services Ltd
18b oshin road,kongi bodija
p.o.box 29460, secretariate,
ibadan.
tel:- 234-2-8105156
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with TE205
Try pri show span 1 and send me the result.
Augustine Olaifa wrote:
Hello,
I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have
my jumper set (i.e closed to use the E1 facility.)
but when i connect the E1 from my telco the LED on the TDM card is
green and also when i look in zttool the status are ok.
when i try to place a call out I get the following:
-- Accepting AUTHENTICATED call from 196.1.178.172:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (g729|gsm),
priority = mine
-- Executing Dial(IAX2/233066391-4, Zap/g5/8105156,30) in new
stack
Apr 21 13:50:43 NOTICE[1565]: app_dial.c:1011 dial_exec_full: Unable
to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
= Everyone is busy/congested at this time (1:0/0/1)
-- Hungup 'IAX2/233066391-6'
-- Accepting AUTHENTICATED call from 196.1.178.172:
requested format = gsm,
requested prefs = (),
actual format = gsm,
host prefs = (g729|gsm),
priority = mine
-- Executing Dial(IAX2/233066391-4, Zap/g5/8105156,30) in new
stack
Apr 21 14:28:52 NOTICE[1095]: app_dial.c:1011 dial_exec_full: Unable
to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
pbx23*CLI
My config files:
zaptel.conf
===
span=1,1,0,ccs,hdb3
span=2,1,0,ccs,hdb3
bchan=1-15,17-30
dchan=16
bchan=31-45,47-60
dchan=46
loadzone = us
zapata.conf
group=5
context=outsideserver
;context= incoming
switchtype=euroisdn
pridialplan=national
priindication=inband
signalling=pri_cpe
resetinterval=60
overlapdial=yes
callprogress=yes
callerid=22004488
;channel=1-15,17-30
bchan=1-15,17-31
dchan=16
txgain=3.0
rxgain=3.0
group=7
context=weareout
;context= incoming
switchtype=euroisdn
;context= incoming
switchtype=euroisdn
pridialplan=national
priindication=inband
signalling=pri_cpe
resetinterval=60
overlapdial=yes
callprogress=yes
callerid=155152
;channel=31-45
;channel=47-61
bchan=31-45,47-60
dchan=46
txgain=3.0
rxgain=3.0
idledial=1551520
[EMAIL PROTECTED]
minunused=2
minidle=1
and this is my dialplan.trying to call out on group 5
extensions.conf
==
[outsideserver]
include= default
include= national
include= local
[weareout]
exten = 1551520,1,Answer
..
..
[national]
exten =_02X.,1,Dial,Zap/g5/${EXTEN:2},30
exten = _0802X.,1,Dial,Zap/g5/${EXTEN},30
exten = _080[36]X.,1,Dial,Zap/g5/${EXTEN},30
;exten = _0804X.,1,Dial,Zap/g5/${EXTEN},30
could this be a hardware problem? or is the TDM card defected?
Please anyone ever had this problem i will like to share how it was
sorted out.
regards
==
Olaifa Augustine
General Data Engineering Services Ltd
18b oshin road,kongi bodija
p.o.box 29460, secretariate,
ibadan.
tel:- 234-2-8105156
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Power over Ethernet (PoE) switch recommendations
Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
D-link has a nice one, optional 5 year warranty on some of the commercial stuff On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote: D-link has a nice one, optional 5 year warranty on some of the commercial stuff Though beware, some of the D-Link ones only have half the ports with PoE. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
Netgear makes a 24 port Layer 3 Managed Switch, with PoE on all 24 ports. It supports both IEEE 802.3af PoE as well as the proprietary Cisco PoE scheme (although the support for Cisco PoE is undocumented). Got one about a year ago for around $1000, which isn't too shaby for a a switch that can do QoS, VLAN and PoE ... has a nice web interface too. Current Netgear model is FSM7326P (not sure if its a new SKU since over a year ago when I got one). http://www.netgear.com/products/details/FSM7326P.php$1000 might seem like a lot, but at ~$42 per port, its in the same ballpark of what you'd pay just for power injectors (not to mention its a layer 3 managed switch). -AdamOn 21/04/06, Andrew Latham [EMAIL PROTECTED] wrote: D-link has a nice one, optional 5 year warranty on some of thecommercial stuffOn 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's.I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office.However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price.I would appreciate any input people have to offer. Thanks, James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better.---___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with TE205
Why do you comment these lines: ;channel=31-45 ;channel=47-61 and put those in the zapata.conf? bchan=31-45,47-60 dchan=46 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MoH issue
Hey everyone,
Hopefully I can describe the problem well enough so bear with me.
There are 3 companies that are tied into our asterisk server. Company A
(us) uses the default settings for music on hold. Companies B and C
however, want something different. For them I have when a call comes
into their dial plan it sets the music on hold to their music and that
seems to work. However, here is the problem. Calling out, it still plays
the old on hold music.
Here is the situation, the 3 companies if they call each other us SIP
and don't even touch the PRI, only outgoing calls outside the companies
will do that. So I also would like if B called C, C's music on hold
would be the one heard.
Here is how I started the dialplan.
[Empire-Outbound]
exten = _.,1,Answer()
exten = _.,n,SetMusicOnHold(OrigMusic)
exten = _.,n,Wait(2)
exten = _.,n,Goto(Empire-Outbound2,${EXTEN},1)
[Empire-Outbound2]
include = A-DirectDial ;Direct Dial Context
include = Empire-VoiceMail ;Voicemail context
include = Empire-Wildcard ;Basic calling function
Does switch between contexts reset the moh? Or can I not change the moh
for SIP channels and only on Zap?
Thanks,
Kevin
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Parallel Dial: Busy detection - stop when any is busy?
Hi All, I'm trying to add this function to my find-me application: when all available numbers are dialed in parallel , if any number is busy, take it at busy and go to voice mail. I read the Dial() Application but there's nothing written about this. My question is, is it possible to do this with Asterisk? Thank you, Pim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with TE205
Hello, I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my jumper set (i.e closed to use the E1 facility.) Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers were completely obsolete now? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with TE205
Jumpers must still be on for E1 mode.RobOn 21/04/06, Remco Barende [EMAIL PROTECTED] wrote: Hello, I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my jumper set (i.e closed to use the E1 facility.)Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers werecompletely obsolete now?___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Definitive list of sounds
Is there a list of sounds (base - as with Asterisk itself, and additional) for the 1.2 release. As in a list with what the content of each file is. There's a list for 1.0.7 on the wiki, but that seems woefully out of date. Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HANGUPCAUSE on SIP channels
Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends 603 Declined. I looked through the source code briefly and it appears that it *should* work. It would be helpful to know if anyone actually uses this feature and if it is working properly for them before we go through with fully debugging and patching this to work for us. Here is the our test extension from extensions.conf: exten = 9218,1,Set(HANGUPCAUSE=1) exten = 9218,2,Hangup According to hangup_cause2sip in chan_sip.c a HANGUPCAUSE of 1 should cause Asterisk to reply to the softswitch with a 404 Not Found SIP message. That doesn't seem to be the case, however. Here is a bit of the verbose console output: (Please note that I added some extra ast_log calls to the source code to generate some extra debugging information.) Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPURI, value=sip:[EMAIL PROTECTED]:5060 Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPDOMAIN, value=192.168.74.254 Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPUSERAGENT, value=PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Apr 21 12:35:18 WARNING[16430]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=SIPCALLID, [EMAIL PROTECTED] -- Executing Set(SIP/nyct-901-539f, HANGUPCAUSE=1) in new stack Apr 21 12:35:18 WARNING[16815]: pbx.c:5983 pbx_builtin_setvar_helper: chan=SIP/nyct-901-539f, name=HANGUPCAUSE, value=1 Apr 21 12:35:18 WARNING[16815]: pbx.c:6057 pbx_builtin_setvar: chan=SIP/nyct-901-539f, name=HANGUPCAUSE, value=1 -- Executing Hangup(SIP/nyct-901-539f, ) in new stack Apr 21 12:35:18 WARNING[16815]: pbx.c:5548 pbx_builtin_hangup: chan-hangupcause=(null) == Spawn extension (nyct, 9218, 2) exited non-zero on 'SIP/nyct-901-539f' Apr 21 12:35:18 WARNING[16815]: chan_sip.c:2471 sip_hangup: ast-hangupcause=16 res=(null) This is all on Asterisk 1.2.7.1. Your line numbers may vary since there were some ast_log lines added. Hopefully this makes some sense to someone. Thanks for any help or input. -- New York Connect Technical Support Staff Eric Futch [EMAIL PROTECTED] (212) 293-2620 Weather for KNYC: Apr 21 11:51a EDT, 59F (15C), Fair, Humidity 49% ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] roundrobin strategy in queues not working as described?
I have set up an operator queue for our receptionist. That way, if she takes a break or is out, by logging out of the queue, calls to the Operator can be handled by other agents. I have set strategy = roundrobin in queues.conf. According to the book ATFoT, roundrobin always starts with the first agent in the queue. This is the desired result. I want all calls to start there, and if she is busy or does not answer, calls should go to the next agent logged into the queue. Yet, I am seeing it behave as if it were rrmemory. I called the Operator while she was not busy, and it rang the other agent. I called again, and this time it came to her. *CLI show queue operator operator has 0 calls (max unlimited) in 'roundrobin' strategy (1s holdtime), W:0, C:1, A:4, SL:0.0% within 0s Members: Agent/200 (Not in use) has taken 1 calls (last was 173 secs ago) Agent/110 (Unavailable) has taken no calls yet Agent/246 (Not in use) has taken 1 calls (last was 302 secs ago) Agent/140 (Unavailable) has taken no calls yet No Callers Any ideas? -- Jim Rice by Design Publishing 11626 N. Tracey Road Hayden, Idaho 83835 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP domain in Asterisk
Hello to all Can someone tell me if its possible to implement a SIP domain with Asterisk (im trying with [EMAIL PROTECTED]). With a SIP domain I mean: -users having URIs with [EMAIL PROTECTED] ( instead of [EMAIL PROTECTED] ) -being able to reach our users anywhere in the world with SIP URIs (and the help of SRV records) -the possibility of dialing [EMAIL PROTECTED] and route the calls through the Internet Can this be done? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] roundrobin strategy in queues not working as described?
Hi Jim, The function roundrobin makes with that asterisk directs the calls for the next free agent, but notorderly. I use the same strategy and functions here very well. Thedifference is that only use the functions ofagent loginokand agent loginoff. I wait to have helped Good luck Regads Josué 2006/4/21, Jim Rice [EMAIL PROTECTED]: I have set up an operator queue for our receptionist.That way, if she takes a break or is out, by logging out of the queue, calls to the Operator can be handled by other agents.I have set strategy = roundrobin in queues.conf.According to the book ATFoT, roundrobin always starts with the firstagent in the queue.This is the desired result.I want all calls to start there, and if she is busy or does not answer, calls should go tothe next agent logged into the queue.Yet, I am seeing it behave as if it were rrmemory.I called the Operator while she was not busy, and it rang the other agent.I called again, and this time it came to her.*CLI show queue operatoroperator has 0 calls (max unlimited) in 'roundrobin' strategy (1sholdtime), W:0, C:1, A:4, SL:0.0% within 0sMembers: Agent/200 (Not in use) has taken 1 calls (last was 173 secs ago) Agent/110 (Unavailable) has taken no calls yet Agent/246 (Not in use) has taken 1 calls (last was 302 secs ago) Agent/140 (Unavailable) has taken no calls yet No CallersAny ideas?--Jim Riceby Design Publishing11626 N. Tracey RoadHayden, Idaho83835___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk FAx-to-Email
-Original Message- From: Wasif [mailto:[EMAIL PROTECTED] Sent: Thursday, April 20, 2006 4:25 PM To: 'asterisk-users@lists.digium.com' Subject: Asterisk FAx-to-Email Hi, I get error when my DID hit to asterisk box which I am using for FAX to Email Service. Sometimes Fax goes through but mostly I get communication error on Fax Machine and on Asterisk I get Comfort noise support incomplete in Asterisk (RFC 3389) error. I am using SIP with G711. My Did provider cannot turn off VAD and Echo from his side, so is there any option or setting I can do at my side to make FAX service more reliable Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to select Ceptral's Voice in Asterisk'sSwift application??
Hi, Check the script. You can assign the voice by -n option, e.g., /opt/swift/bin/swift -n Diane Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shane Young Sent: Friday, April 21, 2006 9:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to select Ceptral's Voice in Asterisk'sSwift application?? Quoting Pimjai Wesnarat [EMAIL PROTECTED]: Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? [cepstral-demo] exten = s,1,Answer exten = s,n,wait(1) exten = s,n,Cepstral(voice name=DuchessHello and welcome to the world of text to speech using Cepstral. My name is Duchess./voice) exten = s,n,Cepstral(voice name=WalterHello and welcome to the world of text to speech using Cepstral. My name is Walter./voice) exten = s,n,Cepstral(voice name=ShoutyHello and welcome to the world of text to speech using Cepstral. My name is Shouty./voice) exten = s,n,Cepstral(voice name=WilliamHello and welcome to the world of text to speech using Cepstral. My name is William./voice) exten = s,n,Cepstral(voice name=WhisperyHello and welcome to the world of text to speech using Cepstral. My name is Whispery./voice) exten = s,n,Cepstral(voice name=RobinHello and welcome to the world of text to speech using Cepstral. My name is Robin./voice) exten = s,n,Cepstral(voice name=LindaHello and welcome to the world of text to speech using Cepstral. My name is Linda./voice) exten = s,n,Cepstral(voice name=EmilyHello and welcome to the world of text to speech using Cepstral. My name is Emily./voice) exten = s,n,Cepstral(voice name=DianeHello and welcome to the world of text to speech using Cepstral. My name is Diane./voice) exten = s,n,Cepstral(voice name=DavidHello and welcome to the world of text to speech using Cepstral. My name is David./voice) exten = s,n,Cepstral(voice name=DuncanHello and welcome to the world of text to speech using Cepstral. My name is Duncan./voice) exten = s,n,Cepstral(voice name=DamienHello and welcome to the world of text to speech using Cepstral. My name is Damien./voice) exten = s,n,Cepstral(voice name=CallieHello and welcome to the world of text to speech using Cepstral. My name is Callie./voice) exten = s,n,Cepstral(voice name=DogHello and welcome to the world of text to speech using Cepstral. My name is Dog./voice) exten = s,n,Cepstral(voice name=AmyHello and welcome to the world of text to speech using Cepstral. My name is Amy./voice) This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with TE205
Weird, I just received a new TE210P card (should be identical only 3.3v) but I cannot find any info on jumper settings om the Digium site? But then again the installation info on the Digium site really sucks. On Fri, 21 Apr 2006, Rob Lith wrote: Jumpers must still be on for E1 mode. Rob On 21/04/06, Remco Barende [EMAIL PROTECTED] wrote: Hello, I am currently running asterisk 1.2.5, and i have a TDM TE205P, i have my jumper set (i.e closed to use the E1 facility.) Does the TE205P use jumpers for T1 / E1 setting? I thought jumpers were completely obsolete now? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk FAX-to-Email
Hi, How can we change the FROM address when Asterisk sends mail (in FAX-to-Email feature). For example it is sending [EMAIL PROTECTED] in FROM address; I need to change it to [EMAIL PROTECTED] Any help? Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom MWI
how about, in sip.conf, [EMAIL PROTECTED] in the [section] for that device? Bill Gibbs wrote: Put your voicemailbox number (usually extension) in the 1.subscribe field. Bill From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Thu 4/20/2006 7:32 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom MWI I have tried everything from voip-info and I still cant get the Polycom 501/601 to display the MWI indicator light. Everything else works just fine. I am using FreePBX set to users and devices mode. Here is the MWI section of the phonexxx.cfg file: mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact msg.mwi.1.callBack=*97 msg.mwi.2.subscribe= msg.mwi.2.callBackMode=disabled msg.mwi.2.callBack= msg.mwi.3.subscribe= msg.mwi.3.callBackMode=disabled msg.mwi.3.callBack= msg.mwi.4.subscribe= msg.mwi.4.callBackMode=disabled msg.mwi.4.callBack= msg.mwi.5.subscribe= msg.mwi.5.callBackMode=disabled msg.mwi.5.callBack= msg.mwi.6.subscribe= msg.mwi.6.callBackMode=disabled msg.mwi.6.callBack=/ /msg i have also tried msg.mwi.1.callBackMode=register Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.techdatapros.com http://www.techdatapros.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] channels change names
Peter Fern wrote: Probably because the Local proxy channel drops out once the two sides have been bridged. If you want the Local chan to stay up, use the /n parameter and the local channel won't perform the native transfer. This does have it's own problems, but should do what you want. Thanks for the tip! Seems to work fine for now. I'll look into something with a variable for the future, but since I'm just getting the information directly from the 'meetme list' command, it would take a bit of rewriting to do that. For now, the quick-and-dirty works perfectly, it seems. -- Jon-o Addleman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call recording
Wai Wu wrote: I notice those options. However, I was looking to start the recording through a third party control program. I know I can do this via chanspy, but is there better way? Not that I know of... I was looking for something kind of similar, and ended up actually using a conference, and adding an ICES streaming channel to it when I want to start recording. Since I'm actually trying to use network streams, this is great for me, but might not be so good for you... -- Jon-o Addleman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best software combination ? Asterisk and Hylafax ?Shall we then allocate destination numbers and or ports for each of those 2 applications ? And if you want to offer to every user, a unique extension for fax and voice, would it still be possible to forward calls from voice application to fax application (for outgoing faxes, the fax application can use its own ressources) ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
SMC 6824MPE.. Does 24 ports POE, with 2x 1GB uplinks, (RJ45 or GBIC) L3 Managed switch. We've got four of them here, and I think they're great, the cost was really reasonable. Ingram no longer lists the MPE model, but it should be available still. Chad -Original Message- Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI in multi-PBX setup
Has anyone tried to set Message Waiting Indicators up when public network access and voicemail service are managed by an Asterisk server TDM-connected to a legacy PBX serving analog and digital phones ?For instance: Location 1:- 200 users on a legacy PBX- among those users, 50 have access to voicemail service- TDM trunk to Location 2Location 2:- 100 users on Asterisk- PSTN access- TDM trunk to Location 1 Does it make any sense to service all 300 users with MWI from the * server ?How does it work ?How could you light legacy PBX phones from Astrisk voicemail ?Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wellgate FXO unit
Anyone know how to set the wellgate unit so incoming calls pass on directly to asterisk? Right now incoming calls ring twice and I hear a recording saying enter the extension. If I go enter the extension it goes on to asterisk just fine. I just want the incoming call to go directly onto asterisk. Anyone found that out? Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI in multi-PBX setup
It really depends on the PBX in use. Avaya uses DTMF tones to light the MWI, you can find examples on the wiki on how to do it. In which case you shouldn't have any problem doing it. Most of the bigger phone systems I have worked with allow the same thru simple DTMF tones. On 4/21/06, Olivier Krief [EMAIL PROTECTED] wrote: Has anyone tried to set Message Waiting Indicators up when public network access and voicemail service are managed by an Asterisk server TDM-connected to a legacy PBX serving analog and digital phones ? For instance: Location 1: - 200 users on a legacy PBX - among those users, 50 have access to voicemail service - TDM trunk to Location 2 Location 2: - 100 users on Asterisk - PSTN access - TDM trunk to Location 1 Does it make any sense to service all 300 users with MWI from the * server ? How does it work ? How could you light legacy PBX phones from Astrisk voicemail ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
On 4/21/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi listers,I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's.I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office.However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price.I would appreciate any input people have to offer. We have been using a 3COM 2226 PWR Plus and have had no issues ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Easier install of QueueMetrics on [EMAIL PROTECTED]
Hello list, we are testing an easier way to install QueueMetrics on an [EMAIL PROTECTED] box (or any other CentOS/RHEL) using the yum package manager. This is still experimental, so it may as well work as not work. We are looking for testers who are willing to try this at home and any feedback would be helpful. This should be it: wget -P /etc/yum.repos.d http://yum.loway.it/loway.repo yum install queuemetrics or a longer version is here: http://astrecipes.net/index.php?n=182 QueueMetrics is a full-blown call center monitoring and reporting system, and is available for free to smaller CCs, home users and individual enthusiasts. Any comment is welcome! l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On 21 Apr 2006, at 18:21, Olivier Krief wrote: To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best software combination ? Asterisk and Hylafax ? Shall we then allocate destination numbers and or ports for each of those 2 applications ? And if you want to offer to every user, a unique extension for fax and voice, would it still be possible to forward calls from voice application to fax application (for outgoing faxes, the fax application can use its own ressources) ? I run the combination of Asterisk and Hylafax, and it is easy for me because I have 3 incoming MSNs. Both Asterisk and Hylafax will see all calls, but Asterisk only has two of the three MSNs configured. The fax number is ignored by Asterisk, so Hylafax answers that after a couple rings. This works perfectly fine, but unfortunately I fell into the DIVA V- BRI doesn't do FAX trap, I bought a V-BRI first. It's now sitting on the shelf, unused. The Asterisk server is using the standard BRI card, which I bought off eBay. IMHO the Eicon website should carry more prominent warnings/ explanations about the lack of FAX capabilities for the V-BRI cards. jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.
This is coming from an * noob. :) I've got two customers, they both are replacing their phone systems with VOIP, and we need to retain both their existing dialplans. One has 5 extensions starting at 100, and the other has 10 extensions, starting at 100. Is there a way to have the same extension number twice in the same asterisk system ? They will have different incoming DIDs of course. I don't want them to be able to see / hear / feel / dial each other internally, either. They must remain completely independent. If anyone's got pointers in a Wiki or PDF somewhere, let me know. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Separating Asterisk SIP extensions from dialing each other.
On 4/21/06, Rick Smith [EMAIL PROTECTED] wrote: This is coming from an * noob. :) I've got two customers, they both are replacing their phone systems with VOIP, and we need to retain both their existing dialplans. One has 5 extensions starting at 100, and the other has 10 extensions, starting at 100. Is there a way to have the same extension number twice in the same asterisk system ? They will have different incoming DIDs of course. I don't want them to be able to see / hear / feel / dial each other internally, either. They must remain completely independent. If anyone's got pointers in a Wiki or PDF somewhere, let me know. Dead easy. Just put them in different contexts. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jingle support - can we test the feature ?
- Original Message - From: Tim Panton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 21, 2006 11:43 AM Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ? On 20 Apr 2006, at 16:39, Robert Rozman wrote: - Original Message - From: Time Bandit [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 20, 2006 4:18 PM Subject: Re: [Asterisk-Users] Jingle support - can we test the feature ? we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Have you looked at Wildfire ? http://www.jivesoftware.org/wildfire/ There is an Asterisk-plugin that update your status automagically when you're on the phone -- Hi, thanks for pointer. I know for that project, but reading about Jingle, Jabber and Asterisk integration it seems not so interesting for me at the moment... Regards, So what aspect of Jingle, Jabber and Asterisk did you mean in your original post ? Well I've read few general interviews and articles about integration of Jingle protocol and Asterisk. There is also IAX version of specification for audio transport. There is asterisk-xmpp effort. The main thing at least in my opinion would be that I could have network of Asterisk servers, and user could use integrated client that would give presence, IM and audio communication in Asterisk compatible way.. So I'm curious if anyone has made and tests or has more info on that Regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parallel Dial: Busy detection - stop when any is busy?
that feature does not exists AFAIK, but you can request it in bugs.digium.com, or offer some money to someone to include it for you. Regards On 4/21/06, Pimjai Wesnarat [EMAIL PROTECTED] wrote: Hi All, I'm trying to add this function to my find-me application: when all available numbers are dialed in parallel , if any number is busy, take it at busy and go to voice mail. I read the Dial() Application but there's nothing written about this. My question is, is it possible to do this with Asterisk? Thank you, Pim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
My favorite one is this one. http://www.provantage.com/d-link-systems-des-1526~7DLNS046.htm On 4/21/06, Steve Kennedy [EMAIL PROTECTED] wrote: On Fri, Apr 21, 2006 at 11:23:16AM -0400, Andrew Latham wrote: D-link has a nice one, optional 5 year warranty on some of the commercial stuff Though beware, some of the D-Link ones only have half the ports with PoE. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering
Hello, I have a problem with one of three [topic] phones. The phone, which is on the LAN in the same subnet as Asterisk, keeps unregistering from the Asterisk server. Whan it is unregistered there is no way to make a phone call from it, but once it is rang by any other of the phones it registers to Asterisk again. The other two are absolutely fine. The problematic one [ecco] puts this messages into messages log file: Apr 21 15:27:23 NOTICE[1099] chan_sip.c: Auto-congesting SIP/ecco-9091 Apr 21 15:27:23 WARNING[1077] channel.c: Avoided initial deadlock for '0x8129f38', 10 retries! sip.conf: [ecco] type=friend username=ecco defaultip=10.10.129.31 host=dynamic nat=no canreinvite=no qualify=yes dtmfmode=rfc2833 Can somebody please explain what these messages mean? What is '0x8129f38'? Can somebody please post working sip.conf (full) for Grandstream (101) phones. I have discussed this topic with Google for quite a long time, but with no results. Thank you very much. Marcel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec problem from SIP to H323
I tried by just upgrading to Ast1.2.4 but same problem. Then I tried to install OH323 but I have this error when compiling :S chan_oh323.c: In function `reload_config': chan_oh323.c:4677: warning: implicit declaration of function `sscanf' chan_oh323.c: At top level: chan_oh323.c:3244: warning: 'update_call_ids' defined but not used gcc -shared -Xlinker -x -g -o chan_oh323.so chan_oh323.o \ -L../wrapper -loh323wrap_s \ -L/usr/src/openh323_v1_17_1/lib -lh323_FreeBSD_x86_r_s \ -L/usr/src/pwlib_v1_9_0/lib -lpt_FreeBSD_x86_r_s \ -lstdc++ -lldap -lldap_r -llber -lpthread -lssl -lcrypto -lexpat /usr/bin/ld: cannot find -lldap gmake[1]: *** [chan_oh323.so] Error 1 gmake[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.3/asterisk-driver' gmake: *** [subdirs_build] Error 1 Does anyone know how to get rid of that? I'm running: OS: FreeBSD 6.0 Asterisk: 1.2.4 OH323: 0.7.3 Thanks in advance. Alejandro. -Mensaje original- De: Oliver Vermeulen [mailto:[EMAIL PROTECTED] Enviado el: Wednesday, April 19, 2006 4:09 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' CC: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Codec problem from SIP to H323 Try to upgrade asterisk to version 1.2.4 Are you using OH323 or H323 ? I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323 and everything worked fine. Cheers, Oliver -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Mejía Evertsz Sent: Thursday, April 20, 2006 12:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Codec problem from SIP to H323 Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to transcode (I don't have licences for g729) - sip.conf has disallow=all allow=g729 - h323.conf has disallow=all allow=g729 The problem: [SIPphone] [sip.conf] [h323.conf] [H323gw] g729---allow=g729 ---allow=g729 ---g729 When I dial to the gateway from the SIPphone using g729 as my sip phone's default codec I get: -- Executing Dial(SIP/amejia-8be1, H323/[EMAIL PROTECTED]) in new stack Apr 19 15:02:14 WARNING[68595]: channel.c:2504 ast_request: No translator path exists for channel type H323 (native 4) to 256 Apr 19 15:02:14 NOTICE[68595]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'H323' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) I don't get it why is it trying to translate anything. There's nothing to translate, cause I'm using g729 in both ends. Well, to make it more interesting, I tried this way: [SIPphone] [sip.conf] [h323.conf] [H323gw] g711---allow=all ---allow=all ---g729 This way, it passes the call to the gateway just giving a waring that it can't find a codec to translate. But at least it passes the call. It rings on the other side, and of course as I don't have any g729 licenses installed it drops the call when answered. -- Executing Dial(SIP/amejia-1fc8, H323/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is making progress passing it to SIP/amejia-1fc8 -- H323/H323gw-2 is ringing -- H323/H323gw-2 answered SIP/amejia-1fc8 Apr 19 15:23:45 WARNING[75484]: channel.c:2685 ast_channel_make_compatible: No path to translate from SIP/amejia-1fc8(4) to H323/H323gw-2(256) Apr 19 15:23:45 WARNING[75484]: app_dial.c:1553 dial_exec_full: Had to drop call because I couldn't make SIP/amejia-1fc8 compatible with H323/H323gw-2 == Spawn extension (test, 444, 1) exited non-zero on 'SIP/amejia-1fc8' Does anybody know how can I get rid of the problem I get on the first scenario? Why does it try to use codec 4 (g711u) if both ends are configured with g729? Please give me some light. I don't know what else to try. Thank you all. Alejandro Mejia ___ --Bandwidth and
Re: [Asterisk-Users] Power over Ethernet (PoE) switch recommendations
We're using Cisco Catalyst 3560 Series 48 port PoE switches. So far, *they just work*. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some EICON Diva 4BRI questions
On Fri, 21 Apr 2006, Olivier Krief wrote: To benefit from DIVA Server 4BRI fax hardware capabilities, what is the best software combination ? Asterisk and Hylafax ? You can use any combination of CAPI based software in parallel. You just need to create the rules for which application shall act on which number/service. Shall we then allocate destination numbers and or ports for each of those 2 applications ? That depends on what you want to do. On incoming calls, each application gets the information about destination number and bearer-capability (service). If you can separate your services with these information, then all is fine. Just configure the applications not to act on the others numbers/services. For Fax and Voice on the same number it is not possible, because the bearer-capability may be the same. So one application to handle both is needed. This can be done with Asterisk/chan-capi. Receiving fax (via CAPI without spandsp) is a feature supported chan-capi if the isdn driver supports it. And if you want to offer to every user, a unique extension for fax and voice, would it still be possible to forward calls from voice application to fax application (for outgoing faxes, the fax application can use its own ressources) ? Outgoing is not a problem, just make sure that one application won't use too many channels and block other appliactions... Having a unique fax extension for each user can be done with Asterisk/chan-capi allone and with another fax-software like Hylafax as well. But if you want to forward a call (which was already accepted by Asterisk) to another CAPI application, it is not possible. (Well, Eicon has a special driver which can do a lot of CAPI extensions, but I did not try this yet). So if you want to do that, I suggest using just chan-capi for receiving faxes and maybe another application for sending faxes. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering
I had a similar problem with a GS101, although with mine, I could make OUTBOUND calls from the phone, but because it wasn't registered, it wouldn't ring if called. I don't know the exact solution, but two things I did was to tell it NOT to subscribe to MWI in the GS config itself, and second, I upgraded the firmware. I actually think the solution was the MWI though. I didn't use the standard TFTP config for the upgrade, but I Don't have the IP addresses I used handy - If you still can't get it to work, let me know, and when I'm home, I'll look up the IP address. -Steve -Original Message- From: Marcel Hecko [mailto:[EMAIL PROTECTED] Sent: Friday, April 21, 2006 2:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream Budge Tone 101 keeps deregistering Hello, I have a problem with one of three [topic] phones. The phone, which is on the LAN in the same subnet as Asterisk, keeps unregistering from the Asterisk server. Whan it is unregistered there is no way to make a phone call from it, but once it is rang by any other of the phones it registers to Asterisk again. The other two are absolutely fine. The problematic one [ecco] puts this messages into messages log file: Apr 21 15:27:23 NOTICE[1099] chan_sip.c: Auto-congesting SIP/ecco-9091 Apr 21 15:27:23 WARNING[1077] channel.c: Avoided initial deadlock for '0x8129f38', 10 retries! sip.conf: [ecco] type=friend username=ecco defaultip=10.10.129.31 host=dynamic nat=no canreinvite=no qualify=yes dtmfmode=rfc2833 Can somebody please explain what these messages mean? What is '0x8129f38'? Can somebody please post working sip.conf (full) for Grandstream (101) phones. I have discussed this topic with Google for quite a long time, but with no results. Thank you very much. Marcel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] still some moh troubles
Hi Bart, If it's anything like the problem we had, you are probably getting what sounds like screeching noises during MOH playback? We had this problem and made it go away by turning off hyperthreading in the server BIOS and starting Linux with noht - this was on a dual Xeon machine. Hope this helps. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Apr 20, 2006, at 6:37 AM, Bart van Daal wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: donderdag 20 april 2006 14:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] still some moh troubles Bart van Daal wrote: Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to Actually, you'll want to use ulaw for Native MOH. CUT #!/bin/sh for filename in *mp3 do eval filename=`echo $filename | cut -f1 -d.` echo Converting $filename sox -V $filename.mp3 -t au -r 8000 -U -b -c 1 $filename.ulaw resample -ql done CUT Doug Thanks for you suggestion Doug, I've converted the files using your script to ulaw but experience the same problem. A thing I forgot to mention is that it only happens on calls passing the trunks to the cisco-routers that terminate to pstn so not on internal sip-sip calls. Normal voice communication runs smoothly over the trunks it's only the moh that causes some problems. again, any pointers like those of Doug are very much appreciated thanks! Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Red Hat AS 4?
Hi Domenico, We're using RHEL 4 ES with no obvious issues Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Apr 21, 2006, at 3:59 AM, Mimmus wrote: Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] server choice
hello I will buy a server to make an IVR solution with asterisk and a te110p T1/E1 digium card. I have two options: 1/ HP Proliant ML370 G4 : Xeon 64bits 3,2Ghz, 1Go Ram, 3 disks SCSI 73Go 2/ Dell PowerEdge 2800: Xeon 64bits 3Ghz, 1Go Ram, 3 disks SCSI 73Go I use linux fedora core 3 and I want a help to choosea goodserver to use with asterisk Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extension match sip address
Is there a way to have an extension match on a sip address? I've tried the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is no good. Is there a better way? -- Jon-o Addleman - http://redowl.dyndns.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] confused about iax and voip providers termination
Hey guys, I'm actively trying to get the big picture on how all this works and relates to each other. I've gone through some basic examples from the book and from the sample files just fine. Now, I've setup an account with a VOIP provider which does IAX termination (exgn.net) After getting an account and following their steps, I can make calls out using my IAX (cubix) and Sip (Xlite) phones. However, I'm a bit confused on the purpose on how my box asterisk box is involved. I completely turned off my Asterisk box, and made a call out using either of my softphones and I was successful. So I gathered that the entire point of iax termination is solely for INBOUND calls TO ME (such if I have a DID). Otherwise I'm just using them as a proxy to forward my sip traffic to them directly from my desktop. I got confused because all references I have seen regarding iax termination and such involved editing your local asterisk box configs as well as the client, but really no clear mention that your config changes only apply to INBOUND calls, and not needed if you want to just make OUTBOUND Sip calls. I want to do BOTH eventually, but since I still have this learning curve, it was just another stumble for me. Do I have the correct picture now? Thanks! Terrelle Shaw ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Definitive list of sounds
Steve Kennedy wrote: Is there a list of sounds (base - as with Asterisk itself, and additional) for the 1.2 release. As in a list with what the content of each file is. There's a list for 1.0.7 on the wiki, but that seems woefully out of date. Any help appreciated. Steve Steve, The format is a little funky, but it should work: http://mirror.astlinux.org/sounds/en-prompts.txt -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
